Re: [Asterisk-Users] AGI Script: calleridnamelookup.agi

2004-12-15 Thread Matt Hohman
Has anyone goten this script to work with other reverse lookup
providers. I also am sitting on a pri line that by standard only gives
number returns.


Thanks,

On Thu, 19 Aug 2004 17:56:16 -0500, Greg Blakely [EMAIL PROTECTED] wrote:
 Is anyone successfully using the AGI script calleridnamelookup.agi  (or
 anything similar) ?
 
 I get both name and number caller ID from my POTS line, but I'd save
 money if I had them deliver ANI only.
 
 I've downloaded and installed the AGI script calleridnamelookup.agi, but
 I always get
 
 -- Executing AGI(SIP/9525485560-5359, calleridnamelookup.agi) in
 new stack
 -- Launched AGI Script
 /var/lib/asterisk/agi-bin/calleridnamelookup.agi
 -- AGI Script calleridnamelookup.agi completed, returning 0
 
 I've even received that result calling in to my iconnect account, which
 delivers only ANI information.
 
 I notice that the URL that it queries does not respond when I enter it
 manually into a browser:
 
 http://www.anywho.com/qry/wp_rl/index.html?npa=719telephone=471.  A
 box comes up that says Fetching Results, and then the request times
 out.
 
 Any idea how to structure the query on ANYWHO or how to use the script
 with another reverse lookup service?
 
 Thanks in advance.
 
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[Asterisk-Users] AB1

2004-09-17 Thread Matt Hohman
We are looking at purchasing a carrier access ab1 channel bank for ptsn interconnect. Has anyone used these before? Any insight?



Thanks,
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
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Re: [Asterisk-Users] AB1

2004-09-17 Thread Matt Hohman
Google? Hey thanks for the info I haven't seen that before. Wonders of modern technology.  It's nice to use the list as a round table and get some insight. 

So they don't have disconnect detection I've heard of people using busy detection is this sufficient am I going to be wishing I paid the extra for the adit 600? 


Thanks for any help,

Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
Email: [EMAIL PROTECTED]
On Sep 17, 2004, at 10:19 AM, Steven Critchfield wrote:

On Fri, 2004-09-17 at 12:13, Matt Hohman wrote:
We are looking at purchasing a carrier access ab1 channel bank forptsn
interconnect. Has anyone used these before? Any insight?

Have you heard about Google? If you put in some terms about what you
want to know, and optionally give it a push in the right direction, it
gives you all kinds of relevant information back. Technology is supposed
to reduce our workload, not make it easier for you to zap the resolve of
those who put forth effort.

http://www.google.com/search?hl=enie=UTF-8q=ab1+site%3Alists.digium.com
6 pages of results. Go read.

-- 
Steven Critchfield [EMAIL PROTECTED]>

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[Asterisk-Users] Dialplan transfer. (h323 transfer)

2004-09-13 Thread Matt Hohman
Uniden's UIP300 transfer key is pretty much just a function key that can be assigned a DTMF value. How could I have asterisk monitor the channel for lets say *#*# then wait for a extension timeout then Start a consultive transfer? Through extensions.conf? Is this even possible? 

Any help would be great!
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
Email: [EMAIL PROTECTED]
On Sep 11, 2004, at 5:47 PM, Matt Hohman wrote:

We have purchased a couple of Uniden UIP300 phones (h.323 only). We love them all features work except hold and transfer. The hold function is local unless you define a DTMF hold sequence... Does Asterisk support this? I had heard that # Was transfer when I hit the pound key I get...  I am sorry that's not a valid extension before i get a chance to enter anything.

Any idea's?

Thanks,
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
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[Asterisk-Users] Dial-plan transfer

2004-09-13 Thread Matt Hohman
Uniden's UIP300 transfer key is pretty much just a function key that can be assigned a DTMF value. How could I have asterisk monitor the channel for lets say *#*# then wait for a extension timeout then Start a consultive transfer? Through extensions.conf? Is this even possible? 

Any help would be great!
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
Email: [EMAIL PROTECTED]
On Sep 11, 2004, at 5:47 PM, Matt Hohman wrote:

We have purchased a couple of Uniden UIP300 phones (h.323 only). We love them all features work except hold and transfer. The hold function is local unless you define a DTMF hold sequence... Does Asterisk support this? I had heard that # Was transfer when I hit the pound key I get...  I am sorry that's not a valid extension before i get a chance to enter anything.

Any idea's?

Thanks,
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
Email: [EMAIL PROTECTED]___
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[Asterisk-Users] h.323 Transfer

2004-09-11 Thread Matt Hohman
We have purchased a couple of Uniden UIP300 phones (h.323 only). We love them all features work except hold and transfer. The hold function is local unless you define a DTMF hold sequence... Does Asterisk support this? I had heard that # Was transfer when I hit the pound key I get...  I am sorry that's not a valid extension before i get a chance to enter anything.

Any idea's?

Thanks,
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
Email: [EMAIL PROTECTED]___
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[Asterisk-Users] UIP300

2004-09-02 Thread Matt Hohman
After having a very good experiance with the UIP200 phones we decided
to purchase a couple of UIP300 (2 line version) Only to releise that
these phones are H.323 not sip. What is the difference between h.323
and sip. What needs to be done to get this phone working with
asterisk?


Thanks,
Matt Hohman
New Heights Church
Vancouver, WA 
360 694 4985
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[Asterisk-Users] uniden Uip300

2004-09-02 Thread Matt Hohman
After having quite a bit of success with the UIP200 we decided to pick up a couple of UIP300 phones. After purchasing them we found that they were H.323 Ip phones. I searched on voip-info.org and have been unable to determine how exactly to setup this phone in asterisk. Any help would be graciously appreciated.
Thanks,
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
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Re: [Asterisk-Users] uniden Uip300 (UIP 200 STATUS)

2004-09-02 Thread Matt Hohman
Yes and no... We had to use the nat=never, We had soem issues with the phone freezing on call waiting this was fixed by the newest firmware 4.59a. But overall a good phone, we just need two lines.

Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
Email: [EMAIL PROTECTED]
On Sep 2, 2004, at 11:29 AM, Steve Maroney wrote:

Sorry to be off topic of your post but did you have the problems
documented on the link below with the UIP200 ?

http://www.voip-info.org/wiki-UIP200

Thank you,
Steve Maroney

On Thu, 2 Sep 2004, Matt Hohman wrote:

After having quite a bit of success with the UIP200 we decided to pick
up a couple of UIP300 phones. After purchasing them we found that they
were H.323 Ip phones. I searched on voip-info.org and have been unable
to determine how exactly to setup this phone in asterisk. Any help
would be graciously appreciated.
Thanks,
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
Email: [EMAIL PROTECTED]

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[Asterisk-Users] Termination Provider

2004-06-25 Thread Matt Hohman
I've been looking for a good iax or sip ==> ptsn provider. Someone with very low cost usa calling and can offer incoming ptsn connections in most markets. The only decent providers I could find were iconnecthere and nufone. Has anyone found someone that really stood out?
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
Email: [EMAIL PROTECTED]

Re: [Asterisk-Users] Swissvoice ip10s

2004-06-24 Thread Matt Hohman
Thanks! well after doing some other .cfg file changes I hardlocked the phone durring startup! Any ideas? (pushing 1,4,7 on powerup isn't helping)


Thanks,
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
Email: [EMAIL PROTECTED]
On Jun 24, 2004, at 12:33 AM, Florian Overkamp wrote:

Hi,

-Original Message-
I've noticed quite a few posts on the list about the swiss 
voice ip10s phone. We recently purchased a few of these 
phones and have had no luck getting the services button to 
work any ideas? are the example .cfg files for this phone? 
any idea when sip firmware is coming? Any help/info would be great.

SIP firmware is currently being tested, there are a few issues that need to
be resolved.

For your MGCP phone: configip10.cfg can be altered to add services:

set features new 1 Transfer NOINFO NOCONF TRUE NOSEQ
set features new 2 Operator NOINFO NOCONF FALSE extension of your
secretary>

And then:

set service_state IDLE NEW 2
set service_state ONE_ACTIVE_LINE NEW 1

This will add two services:

In idle state: An operator button that speeddials your secretary (who can
connect you through ;-)
In conversation: A Transfer button that hookflashes and gives a dialtone
(There were some issues with that, and I have just now been asked by mark to
verify if they have been resolved).

Florian

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Re: [Asterisk-Users] Swissvoice ip10s

2004-06-24 Thread Matt Hohman
Yep I stayed and was able to get through to their ip-phone support in
france. And with me only knowing english and the guy on the other end
speaking broken english we kinda hashed out that it was a bad stick of
flash ram in the phone. Communitech the USA provider for the phone is
overnighting me a new one. AND emailing me the sip firmware for the mgcp
phones.

Thanks,
Matt Hohman
- Original Message - 
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 24, 2004 9:24 AM
Subject: RE: [Asterisk-Users] Swissvoice ip10s


 Hi,

  -Original Message-
  Thanks! well after doing some other .cfg file changes I
  hardlocked the phone durring startup! Any ideas? (pushing
  1,4,7 on powerup isn't helping)

 Ouch! Can you check if it is still fetching any config files from your
 FTP-server at boot ? Might be your configs are corrupted somehow. If it is
 not even doing that, you might just have to ship it back to SwissVoice and
 have them fix it :-P

 Best regards,
 Florian

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[Asterisk-Users] Swissvoice ip10s

2004-06-23 Thread Matt Hohman
I've noticed quite a few posts on the list about the swiss voice ip10s  phone. We recently purchased a few of these phones and have had no luck getting the services button to work any ideas? are the example .cfg files for this phone? any idea when sip firmware is coming? Any help/info would be great.

Thanks,
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
Email: [EMAIL PROTECTED]