Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-09 Thread Wendell Hamilton 
try:
access-list asterisk permit udp any host x.x.x.x eq 1

- Ravichandran Rajagopal [EMAIL PROTECTED] wrote:
 I tried the following ACL command
 
 access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
 2
 
 and I got the following response back
 
 [no] access-list id [line line-num] deny|permit icmp
   sip smask | interface if_name | object-group
 network_obj_grp_id
   dip dmask | interface if_name | object-group
 network_obj_grp_id
   [icmp_type | object-group icmp_type_obj_grp_id]
   [log [disable|default] | [level] [interval secs]]
 Restricted ACLs for route-map use:
 [no] access-list id deny|permit {any | prefix mask | host
 address}
 Command failed
 
 I don't know how to enter into the linux interface of the Cisco Pix
 506
 firewall
 
 
 
 -Original Message-
 From: Joris Cras [mailto:[EMAIL PROTECTED] 
 Sent: Saturday, February 09, 2008 3:23 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
 pix
 506
 
 Ravi,
 
 there is a easy way of creating all those commands in linux.
 just run the following in a shell:
 for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
 permit udp host 192.168.5.0 eq $x any conduit permit udp host;done
 
 This will create all your PIX rules at ones.
  
 I think you could also use Cisco ACL's
  access-list [name] permit udp [source] [destination] range
 This would be in your case something like:
  access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
 10050
 
 Good luck.
 
 Joris
 
 Ravichandran Rajagopal wrote:
  Otis,
  I wanted to clarify what you said and what I comprehended. 
 
  the SIP protocols are disabled in fixup. 
  
  Having said that I guess all I have to do is just the following.
  the inside IP of asterisk server is 192.168.5.0
 
  On the cisco PIX firewall enter the following.
  192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq
 10001 any
  conduit permit udp host
  192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq
 10002 any
  conduit permit udp host
  
  ...
  .
  192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq
 10050 any
  conduit permit udp host
 
  in the rtp.conf in /etc/asterisk 
  change the ending port 2 (which is what it currently is) to
 10050 
 
  Is there an easier way to make the entries in Cisco PIX firewall ?
 
  Thx
  Ravi 
 
  -Original Message-
  From: ListAcct [mailto:[EMAIL PROTECTED] 
  Sent: Saturday, February 09, 2008 12:18 AM
  To: [EMAIL PROTECTED]
  Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: Re: [asterisk-users] oneway audio with asterisk behind
 cisco pix
  506
 
  No problem.  :-P  I thought it might wise to include everything you
 
  needed just in case!! LOL! You are welcome!!!
 
  --Otis 
 
  Ravichandran Rajagopal wrote:

  LOL I guess all I was asking for the changes to be made in the
 Cisco PIX
  506. I think you gave me a short tutorial on VI as well. Thanks
 once
 again
  for this help. Let me work on these changes and test the one-way
 audio
  problem and go from there.
  Thx
  Ravi
 
  -Original Message-
  From: ListAcct [mailto:[EMAIL PROTECTED] 
  Sent: Friday, February 08, 2008 11:55 PM
  To: [EMAIL PROTECTED]
  Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: Re: [asterisk-users] oneway audio with asterisk behind
 cisco pix
  506
 
  Ravi,
 
  I will explain changing the config in asterisk and the pix:
 
  Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port
 span to 
  1 to 10050 (to start, you will need to increase later as ports
 fill
  
  up)

  (use insert to make a change in a file)
 
  to save:
 
 1. esc
 2. shift + colon
 3. wq (to save)
 
  If you made a mistake and do not want to save but you changed
 something 
  in the file:
 
 1. esc
 2. shift + colon
 3. q! (to exit)
 
 
  Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this
 case the 
  static and conduit commands so this is a example from my setup.
 
  Theses are not usable IPs on the Internet or my IPs but just an
  
  example

  outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254)
  dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254)
 
  interface ethernet0 100full (sets the duplex and turns on
 interface)
  interface ethernet1 100full (sets the duplex and turns on
 interface)
 
  nameif ethernet0 outside security0 ( lower security)
  nameif ethernet1 dmz security50 (higher security)
 
  no fixup protocol sip 5060
  no fixup protocol sip udp 5060
 
  ! - this makes things easier so now the pix knows the IP of the
 asterisk 
  box and maps the ip to the name just for configuration purposes
 only so 
  if you had 20

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-09 Thread Wendell Hamilton 
Did you only open up the one port (1)?  You need to open up a range, if 
you're doing it this way, like 1-10020 and then set your rtp ports in 
asterisk to the same range. 

- Ravichandran Rajagopal [EMAIL PROTECTED] wrote:
 I made the following changes and I am still facing one way audio with
 my call flow.
 
 -Original Message-
 From: Wendell Hamilton [mailto:[EMAIL PROTECTED] 
 Sent: Saturday, February 09, 2008 1:58 PM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Cc: Joris Cras; [EMAIL PROTECTED]; Asterisk Users Mailing List -
 Non-Commercial Discussion
 Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
 pix 506
 
 try:
 access-list asterisk permit udp any host x.x.x.x eq 1
 
 - Ravichandran Rajagopal [EMAIL PROTECTED]
 wrote:
  I tried the following ACL command
  
  access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
  2
  
  and I got the following response back
  
  [no] access-list id [line line-num] deny|permit icmp
  sip smask | interface if_name | object-group
  network_obj_grp_id
  dip dmask | interface if_name | object-group
  network_obj_grp_id
  [icmp_type | object-group icmp_type_obj_grp_id]
  [log [disable|default] | [level] [interval secs]]
  Restricted ACLs for route-map use:
  [no] access-list id deny|permit {any | prefix mask | host
  address}
  Command failed
  
  I don't know how to enter into the linux interface of the Cisco Pix
  506
  firewall
  
  
  
  -Original Message-
  From: Joris Cras [mailto:[EMAIL PROTECTED] 
  Sent: Saturday, February 09, 2008 3:23 AM
  To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
 Non-Commercial
  Discussion
  Subject: Re: [asterisk-users] oneway audio with asterisk behind
 cisco
  pix
  506
  
  Ravi,
  
  there is a easy way of creating all those commands in linux.
  just run the following in a shell:
  for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
  permit udp host 192.168.5.0 eq $x any conduit permit udp host;done
  
  This will create all your PIX rules at ones.
   
  I think you could also use Cisco ACL's
   access-list [name] permit udp [source] [destination] range
  This would be in your case something like:
   access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
  10050
  
  Good luck.
  
  Joris
  
  Ravichandran Rajagopal wrote:
   Otis,
   I wanted to clarify what you said and what I comprehended. 
  
   the SIP protocols are disabled in fixup. 
   
   Having said that I guess all I have to do is just the following.
   the inside IP of asterisk server is 192.168.5.0
  
   On the cisco PIX firewall enter the following.
   192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq
  10001 any
   conduit permit udp host
   192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq
  10002 any
   conduit permit udp host
   
   ...
   .
   192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq
  10050 any
   conduit permit udp host
  
   in the rtp.conf in /etc/asterisk 
   change the ending port 2 (which is what it currently is) to
  10050 
  
   Is there an easier way to make the entries in Cisco PIX firewall
 ?
  
   Thx
   Ravi 
  
   -Original Message-
   From: ListAcct [mailto:[EMAIL PROTECTED] 
   Sent: Saturday, February 09, 2008 12:18 AM
   To: [EMAIL PROTECTED]
   Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   Subject: Re: [asterisk-users] oneway audio with asterisk behind
  cisco pix
   506
  
   No problem.  :-P  I thought it might wise to include everything
 you
  
   needed just in case!! LOL! You are welcome!!!
  
   --Otis 
  
   Ravichandran Rajagopal wrote:
 
   LOL I guess all I was asking for the changes to be made in the
  Cisco PIX
   506. I think you gave me a short tutorial on VI as well. Thanks
  once
  again
   for this help. Let me work on these changes and test the one-way
  audio
   problem and go from there.
   Thx
   Ravi
  
   -Original Message-
   From: ListAcct [mailto:[EMAIL PROTECTED] 
   Sent: Friday, February 08, 2008 11:55 PM
   To: [EMAIL PROTECTED]
   Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   Subject: Re: [asterisk-users] oneway audio with asterisk behind
  cisco pix
   506
  
   Ravi,
  
   I will explain changing the config in asterisk and the pix:
  
   Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port
  span to 
   1 to 10050 (to start, you will need to increase later as
 ports
  fill
   
   up)
 
   (use insert to make a change in a file)
  
   to save:
  
  1. esc
  2. shift + colon
  3. wq (to save)
  
   If you made a mistake and do not want to save but you changed
  something 
   in the file:
  
  1. esc
  2. shift + colon
  3. q! (to exit)
  
  
   Cisco Pix

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-08 Thread Wendell Hamilton 
Note also that if you point to the DNS name rather than the IP address of the 
asterisk server on the phones trying to register, you can set NAT=NO on the 
asterisk side and the sip FIXUP command on the PIX will handle everything 
correctly making this workaround unnecessary 


- Original Message - 
From: Ravichandran Rajagopal [EMAIL PROTECTED] 
To: asterisk-users@lists.digium.com 
Sent: Friday, February 8, 2008 8:54:23 PM (GMT-0800) America/Los_Angeles 
Subject: [asterisk-users] oneway audio with asterisk behind cisco pix 506 





Hi, 



I have the Cisco PIX 506 firewall right in front of the asterisk and I am 
getting a one-way audio. I need your help/guidance to resolve this problem. I 
have the “fixups” disabled for SIP in the Cisco PIX 506. Any help rendered by 
you in this subject is greatly appreciated. I have been breaking my head trying 
to resolve this problem for more than one month. I have included the sip.conf 
and the extensions.conf below. 



[SIP.conf] 

; SIP Configuration example for Asterisk 

[general] 

context=incoming 

allowoverlap=no 

bindport=5060 

bindaddr=0.0.0.0 

localnet=192.168.5.0/255.255.255.0 

externip=a.b.ccc.dd 

srvlookup=yes 

allow=ulaw 

allow=alaw 



[incoming] 

type=peer 

nat=no 

canreinvite=no 

host=xx.y.z.aaa 

qualify=yes 

dtmfmode=rfc2833 

context=default 



[extensions.conf] 

[general] 

static=yes 

writeprotect=yes 

clearglobalvars=no 



[default] 

include = customer 

exten = h,1,Hangup 

exten = i,1,Congestion 

exten = i,2,Hangup 



[agnosco] 

include = local-extensions 

include = customer_ivr 

include = incoming 



[customer_ivr] 

include = local-extensions 

exten = s,1,Answer 

exten = s,n,Background(agnosco_intro) 

exten = s,n,WaitExten 



;Dial said extensions 

exten = 5,1,Dial(SIP/[EMAIL PROTECTED],30) 



[incoming] 

exten = 4025901000,1,Goto(1000,1) 

exten = 1000,1,Goto(customer_ivr,s,1) 



Thanks 

sunMoonstar.___
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Re: [asterisk-users] Asterisk behind a PIX firewall?

2007-11-27 Thread Wendell Hamilton 
You can also create the vpn using the existing pix and netgear, eliminating 
more hardware and points of failure. 

- Original Message - 
From: Ricardo Carvalho [EMAIL PROTECTED] 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com 
Sent: Tuesday, November 27, 2007 7:30:35 AM (GMT-0800) America/Los_Angeles 
Subject: Re: [asterisk-users] Asterisk behind a PIX firewall? 

Try to just open port 5060 for SIP signaling on the PIX and also enable the 
INSPECT SIP rule. That way, your PIX firewall will inspect SIP signalling and 
open the necessary UDP ports for the RTP. 

If you have NAT uptream in the network, you should see if in the layer 4 the 
IPs shown in the SIP messages got rewritten by its public IPs, it should have, 
or else you'll never get it working right. 


Regards, 
Ricardo Carvalho. 
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Re: [asterisk-users] Google acquires Grand Central

2007-07-09 Thread Wendell Hamilton 

- Alex Robar [EMAIL PROTECTED] wrote:
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GrandCentral doesn't do anything you can't do with asterisk.  What it does do 
is put those features within reach of an average person by providing a superb 
user interface for the end user, which allows them to self-administer all of 
these wonderful features.  

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RE: [asterisk-users] VoiceXML + Nuance

2007-05-03 Thread wendell hamilton 
I've done considerable work with the voxeo Prophecy platform, and it's
been successful, albeit challenging at times. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Rousse
Sent: Thursday, May 03, 2007 2:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] VoiceXML + Nuance

Hello,

Is there anyone who has ever done a setup of VoiceXML combined with some

licenses from Nuance for the ASR/TTS engine within Asterisk ?
I'm currently working with VoiceGenie, for the VoiceXML + ASR/TTS 
engine, but we are having a couple of issues which I guess are caused by

VoiceGenie.

If there's an alternative, it would be very interesting for us.

Thanks,

-- 
Eric Rousse
System Administrator
514.380.2992
450.655.1001
1.888.641.5800

Telmatik inc.
204 Montarville, suite 250
Boucherville, QC, Canada
J4B 6S2

www.telmatik.com


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This message is confidential. It may also be privileged or otherwise protected 
by work product immunity or other legal rules. If you have received it by 
mistake, please let us know by e-mail reply and delete it from your system; you 
may not copy this message or disclose its contents to anyone. Please send us by 
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RE: [asterisk-users] Shoretel integration into Salesforce.com

2007-03-24 Thread wendell hamilton 
Hello Dean,

We're currently working on improving integration with SugarCRM, with
Salesforce.com integration is next on my list.  FWIW, I've enhanced the
SugarCRM integration to log and screen pop incoming calls identified
by CID, and done a little work to make click to call a bit more robust.
I'm currently working on dynamically routing a call to the correct agent
based on the SugarCRM data.  The goal is to duplicate as much of this
functionality as possible with Salesforce.com after the current project
is complete.  All projects will be released as open-source.

 

Contact me offlist if you'd like more info,

 

Wendell

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
Sent: Saturday, March 24, 2007 7:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Shoretel integration into Salesforce.com

 

http://www.theregister.co.uk/2007/03/24/shortel_sforce 
Shoretel has integrated its IP phone system with Salesforce.com's call
centre software. Using the two together will mean call centre agents get
a reduced admin workload, with automatic call logging and screen pop-ups
with the customer's record.

Is anyone in the asterisk community providing integration into
Salesforce.com?

 

Regards,

Dean Collins
[EMAIL PROTECTED] 
+1-212-203-4357 Ph

  http://click.mexuar.com/webuser/click/7/userurl/Cognation 

www.Mexuar.com http://www.mexuar.com/ 
Want to voice enable your website?
Use Corraleta to reach your customers in 10 seconds or less.

 


This message is confidential. It may also be privileged or otherwise protected 
by work product immunity or other legal rules. If you have received it by 
mistake, please let us know by e-mail reply and delete it from your system; you 
may not copy this message or disclose its contents to anyone. Please send us by 
fax any message containing deadlines as incoming e-mails are not screened for 
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image001.gif
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RE: [asterisk-users] FW: Microsoft buys Tellme

2007-03-16 Thread wendell hamilton 
Have you looked at Voxeo Prophecy? 

 

http://www.voxeo.com/prophecy/

 

cheers,

 

Wendell

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
Sent: Friday, March 16, 2007 10:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] FW: Microsoft buys Tellme

 

http://deancollinsblog.blogspot.com/2007/03/microsoft-buys-tellme.html 

 

I thought I would email this post I made on my blog from yesterday as a
way of stimulating discussion on this. 

 

It looks like the Asterisk community is no closer to getting a Pre-Paid
'Offboard Speech Recognition Processing' SIP gateway than ever. I've
left the http://www.voip-info.org/wiki/index.php?page=Tellme page up for
archival purposes but basically we still have no 'low value' pre-paid
service for the little people.

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb



From: Dean Collins [mailto:[EMAIL PROTECTED] 
Sent: Friday, 16 March 2007 1:11 PM
To: Dean Collins
Subject: [Dean Collins] Microsoft buys Tellme

 


 


 


Microsoft buys Tellme 


Hmmm interesting day today. Basically Microsoft bought Tellme for $800
million dollars.

Long history between myself and Tellme;

So basically in April 2005 someone I knew from Australia went to work
for Tellme in CA. He called to talk to me about his new company and
eventually I was introduced to his boss, TellMe's CTO - Don Jackson.

I was explaining to Don about my involvement with the Asterisk community
and how Tellme could really benefit by getting involved.

What I was trying to suggest they set up was a speech recognition ASP
SIP gateway.
http://www.voip-info.org/wiki/index.php?page=Tellme
http://www.voip-info.org/wiki/index.php?page=Tellme 
or
http://www.Cognation.net/asterisk/tellme
http://www.cognation.net/asterisk/tellme 

The concept behind this is that there were 20,000 Asterisk servers
globally (now probably closer to 35,000), if Tell Me were able to set up
a SIP gateway that allowed the little people to do 'offboard speech
recognition processing' on a low value prepaid basis then there would be
sufficient volume to set up a special channel support team just for the
asterisk community and this whilst initially would require more support,
through code reuse and easy install sample conf files would be a
valuable market.

So long story short, this went back and forward for a while, nothing
came out of it until Tellme announced a joint venture with Skype
http://www.skype.com/partners/voice/
http://www.skype.com/partners/voice/  to do exactly what I was
proposing.

This was just plain dumb because although skype are set up to handle the
front end billing (this is one of the main issues with Tellme setting up
a SIP gateway) no real developer is going to be happy using a skype
channel for their Vxml application.

Of course they've basically made nothing out of it and the idea for a
sip gateway is still sitting there unrealised BUT... thems the breaks.
Besides they get to work for MS now and become zillionaires.

Oh and in case anyone out there wants to tell me there is no money to be
made in servicing the little people check out how many ITSP's exist in
just north america providing termination services
http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Business+North
+America
http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Business+Nort
h+America  the secret is automation from credit card processing to
minute allocation to processing allocation. Get that all sorted and your
done, application stickiness will ensure an ongoing revenue.



Articles linking to the news.
http://gigaom.com/2007/03/14/microsoft-buys-tellme/
http://gigaom.com/2007/03/14/microsoft-buys-tellme/ 

Interesting chart in here showing the VC relationships (all told kicked
in $237M over 8 years to get this payoff)
http://vcratings.thedealblogs.com/2007/03/tellme_the_next_benchmark_and.
php
http://vcratings.thedealblogs.com/2007/03/tellme_the_next_benchmark_and
.php 

http://www.theregister.co.uk/2007/03/14/microsoft_buys_tellme_networks/
http://www.theregister.co.uk/2007/03/14/microsoft_buys_tellme_networks/
 


Cheers,
Dean 

posted by Dean Collins at 11:02 AM
http://deancollinsblog.blogspot.com/2007/03/microsoft-buys-tellme.html
0 comments
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RE: [asterisk-users] FW: Microsoft buys Tellme

2007-03-16 Thread wendell hamilton 
Not their hosted service, their prophecy application (windows based).
It's free for a single port, and reasonably priced for more. No monthly
recurring, just the per port licensing charge.

 

From: Dean Collins [mailto:[EMAIL PROTECTED] 
Sent: Friday, March 16, 2007 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: wendell hamilton
Subject: RE: [asterisk-users] FW: Microsoft buys Tellme

 

Hi Wendel,

Voxeo got in touch with me yesterday however from the initial
discussions it appears they are still looking for a minimum $500+ per
month in billing before they are interested in a relationship.

 

Lets just say I'm in discussions and will keep the list posted as
discussions evolve but as it stands today no it's not viable for the
majority of the Asterisk community.

 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of wendell
hamilton
Sent: Friday, 16 March 2007 4:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] FW: Microsoft buys Tellme

 

Have you looked at Voxeo Prophecy? 

 

http://www.voxeo.com/prophecy/

 

cheers,

 

Wendell

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
Sent: Friday, March 16, 2007 10:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] FW: Microsoft buys Tellme

 

http://deancollinsblog.blogspot.com/2007/03/microsoft-buys-tellme.html 

 

I thought I would email this post I made on my blog from yesterday as a
way of stimulating discussion on this. 

 

It looks like the Asterisk community is no closer to getting a Pre-Paid
'Offboard Speech Recognition Processing' SIP gateway than ever. I've
left the http://www.voip-info.org/wiki/index.php?page=Tellme page up for
archival purposes but basically we still have no 'low value' pre-paid
service for the little people.

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb



From: Dean Collins [mailto:[EMAIL PROTECTED] 
Sent: Friday, 16 March 2007 1:11 PM
To: Dean Collins
Subject: [Dean Collins] Microsoft buys Tellme

 


 


 


Microsoft buys Tellme 


Hmmm interesting day today. Basically Microsoft bought Tellme for $800
million dollars.

Long history between myself and Tellme;

So basically in April 2005 someone I knew from Australia went to work
for Tellme in CA. He called to talk to me about his new company and
eventually I was introduced to his boss, TellMe's CTO - Don Jackson.

I was explaining to Don about my involvement with the Asterisk community
and how Tellme could really benefit by getting involved.

What I was trying to suggest they set up was a speech recognition ASP
SIP gateway.
http://www.voip-info.org/wiki/index.php?page=Tellme
http://www.voip-info.org/wiki/index.php?page=Tellme 
or
http://www.Cognation.net/asterisk/tellme
http://www.cognation.net/asterisk/tellme 

The concept behind this is that there were 20,000 Asterisk servers
globally (now probably closer to 35,000), if Tell Me were able to set up
a SIP gateway that allowed the little people to do 'offboard speech
recognition processing' on a low value prepaid basis then there would be
sufficient volume to set up a special channel support team just for the
asterisk community and this whilst initially would require more support,
through code reuse and easy install sample conf files would be a
valuable market.

So long story short, this went back and forward for a while, nothing
came out of it until Tellme announced a joint venture with Skype
http://www.skype.com/partners/voice/
http://www.skype.com/partners/voice/  to do exactly what I was
proposing.

This was just plain dumb because although skype are set up to handle the
front end billing (this is one of the main issues with Tellme setting up
a SIP gateway) no real developer is going to be happy using a skype
channel for their Vxml application.

Of course they've basically made nothing out of it and the idea for a
sip gateway is still sitting there unrealised BUT... thems the breaks.
Besides they get to work for MS now and become zillionaires.

Oh and in case anyone out there wants to tell me there is no money to be
made in servicing the little people check out how many ITSP's exist in
just north america providing termination services
http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Business+North
+America
http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Business+Nort
h+America  the secret is automation from credit card processing to
minute allocation to processing allocation. Get that all sorted and your
done, application stickiness will ensure an ongoing revenue.



Articles linking to the news.
http://gigaom.com/2007/03/14/microsoft-buys-tellme/
http://gigaom.com/2007

RE: [asterisk-users] Summary of Trixbox vs. custom install

2007-02-18 Thread wendell hamilton 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time
Bandit
Sent: Sunday, February 18, 2007 11:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Summary of Trixbox vs. custom install
snip
 I also include a consideration from mine: I would happily use
 Trixbox, because I did FreePBX setup once and it was a real pain, but
 I'm very frightened by a few issues:

 1) Trixbox Macho installation that installs everything without
 asking. I, for example, would like to use software RAID (maybe it's
 wrong with Asterisk, but I want to do it!). I wouldn't like doing it
 manually after Trixbox installation. I would like to have an
 installer doing it for me. Centos (ex redhat) installer does it, so
 why Trixbox choose to install everything without prompting?
You can just install CentOS with RAID and whatever you want, then use
the Trixbox tar package instead of the ISO. Still, why on earth did
the Trixbox team didn't leave the option of doing a custom install
with the ISO ?
/snip

Please note that the recent (2.x) releases of trixbox allow you to
select which modules to install, including raid.

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RE: [asterisk-users] Is there any Asterisk controllable thermostat?

2006-12-06 Thread wendell hamilton 
I wouldn't IVR this.  I'd do it with extensions.  You have areas,
targets, and actions assigned to an extension digit. So if lights are
appliance type 1, the bedroom is location 1, and the action to turn the
target on is 1, so you dial extension 111 and perform an action based on
the extension.  This also gives you the opportunity to macro events.
Press 539 (sex) and it turns on the Jacuzzi, streams some Tony Bennett
to the speakers, sets the lights to dim, lights the fireplace, porn
begins streaming to the flatscreen and the phone is set to DND. (not
that you'd ever get laid if you were this geeky)




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Prior
Sent: Wednesday, December 06, 2006 7:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Is there any Asterisk controllable
thermostat?

Doug Crompton wrote:
 and it works great. Now I have one more way to control X10 devices. I
can
 even call my VM on the way home and turn on my lights or whatever
before I
 get home.
 
 Doug

I've started to play with writing some code using the Java FastAGI 
interface to connect to my home automation system.  The code is
working and I could now write whatever I wanted, but I haven't figured
out what would be a reasonable menu interface that wouldn't be very
annoying to use.  I'd be very interested to hear what menu structures
and what actual capabilities people have found useful and nice to use.

For example, has anyone come up with something less annoying than the
following dialog:

Press 1 for living room, press 2 for outside, press 3 for bedroom
(I press 2)
Press 1 for porch light, press 2 for garage light
(I press 1)
Press 1 to turn on, Press 2 to turn off, Press 3 to say current status
(I press 1)
congratulations, you just spent several minutes just to turn on a
light!


Steve
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RE: [asterisk-users] Is there any Asterisk controllable thermostat?

2006-12-04 Thread wendell hamilton 
Asterisk can control any x10 capable device.  For a good example, see
http://lorancestinson.blogspot.com/2006/08/asterisk-can-control-world.ht
ml

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan
Zakaria
Sent: Sunday, December 03, 2006 8:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Is there any Asterisk controllable thermostat?

 

I am wondering if there is any such thermostat available which can be
controlled from Asterisk. Like you call your home pbx, dial some
extension, e.g. 333 and it asks to set the temperature, you enter a
temperature, and it sets the thermostat to that temperature. This
thermostat will be very useful, e.g. when you're coming back home after
a few days and now its snowing and you want home to be warm on your
arrival, you can turn the furnace on an hour before your arrival.

Is there any such thermostat available, and for that matter any other
Asterisk controllable home automation devices? 

-- 
Zeeshan A Zakaria 


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RE: [asterisk-users] Blind transfer 3/4 digits

2006-09-04 Thread wendell hamilton 
Please excuse the top-posting.

In features.conf, uncomment transferdigittimeout and adjust its timing as 
desired.  You may also want to uncomment and adjust featuredigittimeout to a 
higher value as well.  Also, since the dialplan does first match, you can 
eliminate the problem by putting the 4 digit extensions before the 3 digit 
extensions in the dialplan.

See the match as you go section at
http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching

HTH

routerguy

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger
Sent: Monday, September 04, 2006 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind transfer 3/4 digits

David Gagnon wrote:
 Ronald,

   Like someone already told you, you should explain more clearly the
 way you try to transfer, we need more details on the procedure, using which
 button on which phone. We need every detail to help you. This as nothing to
 do with the way the dial plan is loaded, this is totally false.

   I'm sure most of the people here don't understand how you try to
 transfer.

   David

   

David,

I am not sure how the explanation how to punch the keys changes 
something,  ;-.)

Ok, here we go:

Snom:
pick up the phone and hit ##6014 followed by [ok]

Noname:
pick up the phone and hit ##6014 must be pushed very fast!!! No end 
# needed, since the phone 601 starts to ring as soon I reach 1.

In my opinion Asterisk remembers all numbers and therefore it does not 
wait for the 4, since it found a match. This is in VoIP (in my opinion) 
wrong, since overlapping numbers are allowed.

Sip message / dtmf, this is something different! How is the transfer 
made? Maybe snom does send a sip message, while the noname only send 
dtmf tones.

bye

Ronald



 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De la part de Ronald
 Wiplinger
 Envoyé : 4 septembre 2006 09:22
 À : Asterisk Users Mailing List - Non-Commercial Discussion
 Objet : Re: [asterisk-users] Blind transfer 3/4 digits

 Koopmann, Jan-Peter wrote:
   
 On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:

   
 
 try that way. However, I have doubts as well. If you are right, than
 why snom phone does not have this problem? Would not here also the
 first match count?   
 
   
 Because the transfer button on the SNOM is using a totally different
 
 mechanism than sending # to Asterisk. On your snom configuration (like ours)
 the phone does not start to create/send a SIP message until you hit OK. At
 that time the entire number is there and a complete SIP transfer is created.
 Cool down a bit. The problem you are having is most probably just a dialplan
 problem. It takes some time and experience to get those things right. No
 need to yell here...
   
   
 
 What's happen to you guys? I am not yelling, just asking.
 It is sure not a dialplan question! If it would be a dialplan question, 
 than it would be for each dialing, but it isn't.

 You mentioned SIP message and that makes me wonder! Are we not using 
 here dtmf ?? that is in my opinion not a sip message, isn't it?
 If it is a sequence of tones, than why is it different if it is in a 
 string (like snom) or another phone, with single tones?
 If we understand this part, than is the question, where can I turn on 
 the system to take a longer break between tones still as a string?

 Back to the dialplan:
 A Voip number can have different length of digits. Each number is seen 
 as a complete picture, and so a three digit and a four digit number is 
 something different. While in the legacy telephony the digits are worked 
 down one by one and if there is no more use of the digits, they are just 
 garbage and will be not used. Unlike in VoIP, where you can have a three 
 digit number and if you dial four digit, than it is a WRONG number  
 I just verified that: I dialed from 601 to  61522, however, 61522 does 
 not exist, but 615 exists. Guess what? I get a busy tone! That should 
 proof my thoughts (and that without yelling, ... hehehehe)

 bye

 Ronald
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 ---
 avast! Antivirus: Inbound message clean.
 Virus Database (VPS): 0635-4, 2006/09/01
 Tested on: 2006/9/4 ¤U¤È 11:40:24
 avast! - copyright (c) 1988-2006 ALWIL Software.
 http://www.avast.com




   


-- 
Ronald Wiplinger  (CEO of ELMIT)
http://www.elmit.com  http://voip.elmit.com  http://e-paper.elmit.com 
Tel. (M) +886.939.775.516  (O)

RE: [asterisk-users] Unknown CLI output

2006-08-30 Thread wendell hamilton 
I'm experiencing the same CLI messages.  No Linksys, a couple of IAXY's,
a couple of polycom's, various softphones, no zap at all.  I'm not
experiencing any noticeable problems, just the cli messages.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos
Leal
Sent: Wednesday, August 30, 2006 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unknown CLI output

OK, changed the register interval for the Linksys PAP2T to 10 times  
longer and the output described earlier on the CLI also appears to  
follow the same schedule.

I guess I'll have to check a Linksys list to see what could be  
causing this and if I should expect things to get worse.


On Aug 29, 2006, at 8:21 PM, Carlos Leal wrote:

 I'm wondering if anyone can tell me what the following output,  
 repeated about once per minute on my verbose=5  CLI , means.

 -- Contact header: transport
 -- Contact header: q
 -- Contact header: transport
 -- Contact header: q

 I'm on the latest version, 1.2.11, and am recovering from a too- 
 near lighting strike that caused damage here and there. Asterisk is  
 back up, minus a clone FXO card that the phone company said was  
 causing a short in the phone line.  SIP and IAX lines seem to work  
 normally again except for this message that pops up about once a  
 minute.

 Could it be a PAP2T that refreshes registration every 60 seconds?  
 If so, what's changed?

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RE: [Asterisk-Users] Orative

2006-04-17 Thread wendell hamilton 
And they have better sushi than Montana and Wyoming

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew D
Kirch
Sent: Monday, April 17, 2006 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Orative

The weather isn't as good in Indiana.

Douglas Garstang wrote:
 Jeez. Why does every startup in the universe have to be in the bay 
 area. :(

 -Original Message-
 *From:* Dean Collins [mailto:[EMAIL PROTECTED]
 *Sent:* Monday, April 17, 2006 1:00 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [Asterisk-Users] Orative

 Has anyone heard anything about these guys? Anyone seen anything
 like this?

 http://www.orative.com/solutions.php

 It's seems very cool, basically uses GPRS as a digital overlay on
 your mobile phone for additional functionality such as presence
 and IM though I'm sure they have some other functionality
 (voicemail access, call announce etc) coming down the pipeline.

 Any thoughts, how hard would it be to build something like this
 from scratch for the asterisk platform?

 Regards,

 Dean Collins

 Cognation Pty Ltd

 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]

 +1-212-203-4357

 +61-2-9016-5642 (Sydney in-dial).




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RE: [Asterisk-Users] Instant Message?

2006-04-09 Thread wendell hamilton 
I have Jive (wildfire) 2.4.4 running on a win2k3 server box with the
asterisk plugin.  Installed without significant problems, has been up
and running for about 6 wks now.  Conference rooms especially are
convenient.  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry
Garrison
Sent: Sunday, April 09, 2006 7:48 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Instant Message?

I tried the latest version of Jive over the weekend and I have to say it
is
a giant pile of crap. I did this on multiple machines on both Linux and
Windows, and after setting everything up, the moment you add the
asterisk
module, all authentication and user setup is lost and there is no way to
log
back in as the admin to fix it. If anyone has any more positive
experience I
would like to hear about it as it sounds very interesting.
-Kerry
 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Steve Totaro
 Sent: Sunday, April 09, 2006 6:40 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Instant Message?
 
 Zhiqiang Li wrote:
  Hi all,
   
  My client softphone supports IM feature. Does any warmheated expert 
  know if Asterisk can support IM also at server side? If so, 
 is there 
  any related documents or weblinks?
 
  --
  Thanks  Best Regards!
 
  Steven Li
 

 
 I am not sure exactly what you are trying to do but Jive 
 Messenger has asterisk add-ons and functionality.  Might be 
 worth a look for ya. 
 
 Thanks,
 Steve Totaro
 
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RE: [Asterisk-Users] Telasip

2006-04-06 Thread wendell hamilton 
I've had the same excellent responsiveness from telasip, on the rare
occasion that I've had issues.  
YMMV


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Thursday, April 06, 2006 6:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Telasip

then you got lucky.

On 4/6/06, Jeremy [EMAIL PROTECTED] wrote:
 I was wrong, it was me having the problems. I must say I found their 
 support very helpful, I reached a REAL PERSON with one try.  I know a 
 lot of people slam their providers on this list. . . Can anyone tell 
 us their good experiences?

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
 Sent: Thursday, April 06, 2006 8:38 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Telasip

 What type of problem?  They made some drastic changes and didn't let 
 people know.

 On 4/6/06, Jeremy [EMAIL PROTECTED] wrote:
 
  Anyone having any problems with TelaSip today?
 
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RE: [Asterisk-Users] Coice recognition IVR?

2006-04-03 Thread wendell hamilton 
Voxeo has a new product (still beta) called prophecy2006.  Free for 2
ports, inexpensive for more.  I've tested and implemented with asterisk
and that works nicely.  Supports CallXML, CCXML, and VXML.




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cosmin
Prund
Sent: Monday, April 03, 2006 9:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Coice recognition IVR?

Unfortunately I already gave up myself!

At first glance setting up Sphinx looks like a real pain and, while my
threshold for such pain would definitively allow me to work with it, my
available time can't support this. And I am sorry, because it would
look really nice talking to your box, asking it to reboot or
something. Very star-trek

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users- 
 [EMAIL PROTECTED] On Behalf Of Joshua Colp
 Sent: Monday, April 03, 2006 7:06 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Coice recognition IVR?
 
 Cosmin Prund wrote:
  Hello everyone.
 
  Is it possible to do some very basic voice recognition from within 
  Asterisk's dialplan? What I'm aiming at is the ability to speak the
 digits I
  want to dial from my mobile phone. Dialing digits on my mobile phone
 while
  driving is not all that safe...
 
  Thanks for any input,
  Cosmin Prund
 
 
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 This has been discussed a lot before, and people usually end up giving

 Sphinx a go and seeing how it is. If you search the mailing list 
 archives you might find something useful.
 
 Joshua Colp
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RE: [Asterisk-Users] Any new Voice Recognition devs?

2006-03-29 Thread wendell hamilton 








Not open source, but free for 2 ports:  Prophecy2006 by Voxeo. 
Supports CCXML, CallXML and VXML.  Interops nicely w/ asterisk.









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Neil
Skowronek
Sent: Wednesday, March 29, 2006 11:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Any new Voice Recognition devs?



Hi list,

Havent seen any recent postings regarding Voice Recognition

I checked out Sphinx a few months ago but was wondering if anyone has had any
recent experiences, success/failure stories with any new or updated Open Source
VR software as of late (including Spyinx). 

I'm hoping to make the enter name, number and address process better.

THX

Neil







Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone
calls. Great
rates starting at 1¢/min.





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RE: [Asterisk-Users] Bluetooth headsetin handsfree modewith SJPhoneorX-lite

2006-03-28 Thread wendell hamilton 
I didn't change it. I just had a heckofa time getting it to load.
Unfortunately, I neglected to document the process, since this was on my
home PC and I wasn't planning on replicating it. If I can find some
time, I'll see if I can recreate the process.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Tuesday, March 28, 2006 5:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bluetooth headsetin handsfree modewith
SJPhoneorX-lite

Hi,

This is interesting you used the ctp profile. Did you document how you
changed it?

Thanks

wendell hamilton wrote:
 I am able to pickup, hangup, and flash, using the buttons on the phone

 with all of the soft clients and both of the headsets I mentioned
below.
 I don't believe I had to change anything on the client side, just had 
 to get the ctp (cordless telephone profile) working in the bluetooth 
 stack, which was a pita. I struggled with the same issue...I could use

 the headset, but that's not very handy when you have to run to the pc 
 to pickup or hangup a call!  I'm working with doing a 
 voice-recognition dial using the new voxeo prophecy server to add 
 functionality to this, so that I can outgoing dial as well.
   
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RE: [Asterisk-Users] Bluetooth headset in handsfree mode with SJPhoneor X-lite

2006-03-27 Thread wendell hamilton 
Try replacing the XP Bluetooth stack with the widcomm drivers...google
is your friend!


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Monday, March 27, 2006 6:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Bluetooth headset in handsfree mode with
SJPhoneor X-lite

Hi,

After much searching I have found that it might be possible to get a 
bluetooth headset to answer/hangup with SJPhone or Xlite if the headset 
supports handsfree mode. My Toshiba bluetooth stack supports this but I 
have not been able to figure out how to enable it. Also Windows XP 
desktop bluetooth stack does not support handsfree but Windows CE does 
(go figure). Has anyone got handsfree mode working with a bluetooth 
headset? How about working with SJPhone or Xlite or some other SIP 
phone? For some reason the SJPhone when used with a bluetooth headset 
disconnects/reconnects bluetooth when the answer/hangup button is used 
on the headset (how the hell did that come about). Using a bluetooth 
headset with a SIP phone and asterisk would really help me by removing 
those pesky wires

Thanks
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RE: [Asterisk-Users] Bluetooth headset in handsfree modewith SJPhoneor X-lite

2006-03-27 Thread wendell hamilton 
Hi,

You need to have completely replaced the Microsoft driver, because it
doesn't support the headset or ctp Bluetooth profiles.  This gave me
fits!  I followed the instructions at
http://www.windowsdevcenter.com/pub/a/windows/2005/07/05/bluetooth.html
and it works with both a Plantronics and a Motorola Headset, and I can
answer calls with idefisk, eyebeam, x-lite, and kapanga. 

If you end up not having both of these in the Bluetooth service
selection, you won't end up with the results you're looking for.

HTH 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Monday, March 27, 2006 9:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bluetooth headset in handsfree modewith
SJPhoneor X-lite

Hi,

I am not having trouble with the bluetooth stack since the Toshiba stack

has the headset profile which supports a subset of AT commands 
http://en.wikipedia.org/wiki/AT_command from GSM 07.07 for minimal 
controls including the ability to ring, answer a call, hang up and 
adjust the volume. The problem is getting the softphone to work with 
these AT commands so that the answer/hangup function will work from the 
bluetooth headset.

Thanks

wendell hamilton wrote:
 Try replacing the XP Bluetooth stack with the widcomm drivers...google
 is your friend!


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Chuck
Bunn
 Sent: Monday, March 27, 2006 6:21 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Bluetooth headset in handsfree mode with
 SJPhoneor X-lite

 Hi,

 After much searching I have found that it might be possible to get a 
 bluetooth headset to answer/hangup with SJPhone or Xlite if the
headset 
 supports handsfree mode. My Toshiba bluetooth stack supports this but
I 
 have not been able to figure out how to enable it. Also Windows XP 
 desktop bluetooth stack does not support handsfree but Windows CE does

 (go figure). Has anyone got handsfree mode working with a bluetooth 
 headset? How about working with SJPhone or Xlite or some other SIP 
 phone? For some reason the SJPhone when used with a bluetooth headset 
 disconnects/reconnects bluetooth when the answer/hangup button is used

 on the headset (how the hell did that come about). Using a bluetooth 
 headset with a SIP phone and asterisk would really help me by removing

 those pesky wires

 Thanks
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RE: [Asterisk-Users] Bluetooth headset in handsfree modewith SJPhoneorX-lite

2006-03-27 Thread wendell hamilton 
I am able to pickup, hangup, and flash, using the buttons on the phone
with all of the soft clients and both of the headsets I mentioned below.
I don't believe I had to change anything on the client side, just had to
get the ctp (cordless telephone profile) working in the bluetooth stack,
which was a pita. I struggled with the same issue...I could use the
headset, but that's not very handy when you have to run to the pc to
pickup or hangup a call!  I'm working with doing a voice-recognition
dial using the new voxeo prophecy server to add functionality to this,
so that I can outgoing dial as well.  

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Monday, March 27, 2006 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bluetooth headset in handsfree modewith
SJPhoneorX-lite

Hi,

Are you able to answer calls by pressing the answer/hangup button on
your headset or are you using the computer to answer the calls with a
SIP phone. I can get the head set to work fine with the PC and sip
phones what I cannot do is get the answer/hangup button on the head set
to actually answer a call (for some odd reason it turns the bluetooth on
and off and this occurrs with several different head sets).

Thanks for the info on the stack - good reference

wendell hamilton wrote:
 Hi,

 You need to have completely replaced the Microsoft driver, because it 
 doesn't support the headset or ctp Bluetooth profiles.  This gave me 
 fits!  I followed the instructions at 
 http://www.windowsdevcenter.com/pub/a/windows/2005/07/05/bluetooth.htm
 l and it works with both a Plantronics and a Motorola Headset, and I 
 can answer calls with idefisk, eyebeam, x-lite, and kapanga.

 If you end up not having both of these in the Bluetooth service 
 selection, you won't end up with the results you're looking for.

 HTH

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Chuck 
 Bunn
 Sent: Monday, March 27, 2006 9:46 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Bluetooth headset in handsfree modewith 
 SJPhoneor X-lite

   

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RE: [Asterisk-Users] Info about F1000G

2006-03-02 Thread wendell hamilton 
Ditto on the volume issue, it's EXTREMELY low, but the latest firmware does 
allow login to web-portal protected billed wifi systems.  

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vahan Yerkanian
Sent: Thursday, March 02, 2006 12:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Info about F1000G

Tomislav Parčina wrote:
 Does anybody use UTStarcom F1000G Wi-FI VoIP phone?
 http://www.utstar.com/Solutions/Handsets/WiFi/
 
 I'm planning to buy one and I need to know did you have any problems with 
 phone. What is the sound quality? How close you need to be to the access 
 point?
 
 Please, any information's are useful to me.

Bought one as a sample. Build quality is above average. Battery life is 
very short. Maximum volume of the headpiece is very very low, e.g. you 
can't hear anything while talking on the street or in a noisy 
environment like a bar, etc. Menu is not user friendly, sip and network 
settings are too far from each other. Doesn't support wifi networks that 
use 802.1x billing systems.

Resume: OK phone for home / small office use, but you'll always want to 
get a better one. Has become a member of Hall of Junk for our HQ 
museum. ;)

HTH,
Vahan

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[Asterisk-Users] sip registration fails with 404

2006-02-23 Thread wendell hamilton 








Can anyone give me any
direction as to why I'm getting a 404 during the registration process. Sip
Debug is:



-- SIP read from
192.168.99.110:5060:

REGISTER
sip:asterisk1.rightsolve.com SIP/2.0

Via: SIP/2.0/UDP
192.168.99.110:5060;branch=123456789

To:
sip:[EMAIL PROTECTED]

From:
sip:[EMAIL PROTECTED];tag=12345

CSeq: 1 REGISTER

Call-ID:
f97f33fb6c6a82c2efc0a16258e93d6b

Max-Forwards: 70

User-Agent: VCS

Contact:
sip:[EMAIL PROTECTED]

Expires: 600

Content-Length: 0





--- (11 headers 0 lines)---

Using latest REGISTER
request as basis request Sending to 192.168.99.110 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.99.110:5060:

SIP/2.0 404 Not found

Via: SIP/2.0/UDP
192.168.99.110:5060;branch=123456789;received=192.168.99.110

From:
sip:[EMAIL PROTECTED];tag=12345

To:
sip:[EMAIL PROTECTED];tag=as5106c249

Call-ID:
f97f33fb6c6a82c2efc0a16258e93d6b

CSeq: 1 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Max-Forwards: 70

Contact:
sip:[EMAIL PROTECTED]

Content-Length: 0



TIA,



routerguy







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