from:"Ferguson, Michael"

Hi Guys

I too am trying to do exactly the same thing in being a provider for family 
members. My Asterisk server is on a public ip, my home is behind a Watchguard 
Firebox, my job is also behind a Firebox. I am using a combination of Cisco 
7960, Linksys 941 and XTEN Softphone. Sometimes it works and sometimes it does 
not.

You idea on using a IAX2 softphone appears to be what will solve my problem.

Thanks very much Post more ideas. 'preciate it.





-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Ellson
Sent: Thursday, September 07, 2006 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.


Bruce,

I *just* tested the XtremePhone, IAX2 softphone. Other than trying to figure 
out how to get it to send proper CallerID to the other phones, it worked right 
off, in both directions. Excellent!

Perhaps working the IAX2 angle will be less of a hassle, I will go looking for 
one that does video now.

Maybe it's time to buy an IAX2-ATA adaptor and see how well that works over the 
net.

Nick

As for the SIP logs, I start Asterisk with -c already, I did a sip debug 
and tried my call from the house to my remote SIP phone. YIKES!! 
Gunna take a bit to understand all that, but I think I did see an INVITE, and a 
CANCEL twice in a row and I did not hit the hang-up switch. So that might 
explain why no connection is made, and the called gets my voice-mail (according 
to my wife)



--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.


On Thu, 7 Sep 2006, Bruce Reeves wrote:

 Nick,

 I have done what you are talking about as far as being a provider for family
 members. I used an IAX softphone mainly to eliminate the need for so many
 holes in the firewall. And secondly because the idefisk IAX softphone
 allowed me to extract the zip version, configure the phone, and zip the
 folder up and email it to my family members. So for my mom it was simply
 unzip the folder and

 On 9/7/06, Nick Ellson [EMAIL PROTECTED] wrote:
 

  Bob,

  I will up the logs today, have my phone at work with me. (though the Wife
  and Kids are not up yet ;)

  Anything specific I should target?
 

  Nick
 

  --
  Nick Ellson
  CCDA, CCNP, CCSP, CCAI,
  MCSE 2000, Security+, Network+
  Network Hobbyist, VFR Private Pilot.
 

  On Thu, 7 Sep 2006, Bob Chiodini wrote:
 
   Nick,
  
   Anything helpful in the asterisk or system logs.
  
   Try bumping up the debug and verbose levels see what shows up on the
   console.
  
   Weird that it would work inbound and not outbound.
  
   Bob...
  
  
   On Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote:
   
Hey all,
   
A previous annoyance with not being able to call out to my brother on
  FWD
from my Asterisk system had me thinking that since I have my own PBX,
  and
that system has it's own 1-to-1 static NAT to the internet, I should 
be
 
able to act as the provider for him or any of my family, and have them
  as
local extensions of my PBX, right?
   
So I took my laptop to work (using the X-Lite SIP softphone) and watch
  my
ACL logs on my router for any denies to my Asterisk box. As expected
udp/5060, then once that was open, a series of randomish udp/1+
requests. My phone registered, and I tried to call one of the phones
behind a PAP2. Worked first shot, and just as clear and responsive as
  it
was when I was home. But, the phones at home could not call me, they
  when
to voice mail.
   
I had heard that SIP doesn't survive NAT all that well, and that IAX
native phones do a better job. My question is, given my description of
  how
I am set up and what I am trying to accomplish, should I be looking at
  SIP
or is IAX a more robust choice? (I was hoping to get video working as
well, h.263 I believe it is).
   
Nick
   
   
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 -- 
 Bruce
 Nortex Networks


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RE: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.




Thanks but question!

In this folder I see:
the original Zip file i downloaded - 
idefisk137.zip
addressbook.conf
idefisk.conf
hostory.txt
iaxclient.dll
Idefiskmanual.htm
idefisk.exe

Using Wordpad, I opened addressbook.conf and 
idefisk.conf but saw no reference to the IP address of my Asterisk server. Where 
is this info included in the zip file you sent or did you folks have to do the 
actual config of the softphone?

Thanks again


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce 
ReevesSent: Thursday, September 07, 2006 1:46 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[asterisk-users] Softphones IAX vs. SIP, remote 
connectivity.
Micheal,I do this with the zip version of idefisk avaliable 
here : http://asteriskguru.com/tools/idefisk_windows.phpI 
download and extract the files the run the phone and configure the settings and 
the speed dials, all of which is stored in the folder with the application. I 
then zip it up and email it with instructions to unzip and run the program. 
Works great on my thumb drive also. 
On 9/7/06, Ferguson, 
Michael [EMAIL PROTECTED] wrote:

  
  
  Bruce,
  
  How do you 
  go about accomplishing configuring the phone, zipping it up and sending it 
  over to your family?
  
  Thanks
  
  
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of 
  Bruce ReevesSent: Thursday, September 07, 2006 8:37 
  AM
  To: Asterisk Users 
  Mailing List - Non-Commercial DiscussionSubject: Re: 
  [asterisk-users] Softphones IAX vs. SIP, remote 
  connectivity.
  
  
  Nick,I have done what you are talking about as far as being 
  a provider for family members. I used an IAX softphone mainly to eliminate the 
  need for so many holes in the firewall. And secondly because the idefisk IAX 
  softphone allowed me to extract the zip version, configure the phone, and zip 
  the folder up and email it to my family members. So for my mom it was simply 
  unzip the folder and 
  On 9/7/06, Nick 
  Ellson [EMAIL PROTECTED]  
  wrote: 
  Bob,I 
will up the logs today, have my phone at work with me. (though the 
Wifeand Kids are not up yet ;)Anything specific I should 
target?Nick--Nick EllsonCCDA, CCNP, CCSP, 
CCAI, MCSE 2000, Security+, Network+Network Hobbyist, VFR Private 
Pilot.On Thu, 7 Sep 2006, Bob Chiodini wrote: 
Nick, Anything helpful in the asterisk or system 
logs. Try bumping up the debug and verbose levels see what 
shows up on the  console. Weird that it would work 
inbound and not outbound. Bob... On 
Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote: 
Hey all, A previous annoyance with not being able to 
call out to my brother on FWD from my Asterisk system had me 
thinking that since I have my own PBX, and that system has it's 
own 1-to-1 static NAT to the internet, I should be  able to act 
as the provider for him or any of my family, and have them as 
local extensions of my PBX, right? So I took my 
laptop to work (using the X-Lite SIP softphone) and watch my  
ACL logs on my router for any denies to my Asterisk box. As 
expected udp/5060, then once that was open, a series of 
randomish udp/1+ requests. My phone registered, and I tried 
to call one of the phones  behind a PAP2. Worked first shot, and 
just as clear and responsive as it was when I was home. But, the 
phones at home could not call me, they when to voice 
mail. I had heard that SIP doesn't survive NAT all 
that well, and that IAX  native phones do a better job. My 
question is, given my description of how I am set up and what I 
am trying to accomplish, should I be looking at SIP or is IAX a 
more robust choice? (I was hoping to get video working as  well, 
h.263 I believe it is). 
Nick 
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visit: http://lists.digium.com/mailman/listinfo/asterisk-users 
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and Colocation provided by Easynews.com 
--asterisk-users mailing listTo UNSUBSCRIBE or update options 
visit:  http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks 
  ___--Bandwidth 
  and Colocation provided by Easynews.com -- 
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  visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks 
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RE: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.

Great. Thanks very much 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Ellson
Sent: Thursday, September 07, 2006 2:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.

You need to MAKE a sample config by configuring your phone first, then ya get a 
nice little .xml config file you can batch tweak. :) That's what I found out.

--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.

On Thu, 7 Sep 2006, Ferguson, Michael wrote:

 Thanks but question!

 In this folder I see:
 the original Zip file i downloaded - idefisk137.zip
 addressbook.conf
 idefisk.conf
 hostory.txt
 iaxclient.dll
 Idefiskmanual.htm
 idefisk.exe

 Using Wordpad, I opened addressbook.conf and idefisk.conf but saw no 
 reference to the IP address of my Asterisk server. Where is this info 
 included in the zip file you sent or did you folks have to do the actual 
 config of the softphone?

 Thanks again

 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
 Sent: Thursday, September 07, 2006 1:46 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.

 Micheal,

 I do this with the zip version of idefisk avaliable here : 
 http://asteriskguru.com/tools/idefisk_windows.php

 I download and extract the files the run the phone and configure the settings 
 and the speed dials, all of which is stored in the folder with the 
 application. I then zip it up and email it with instructions to unzip and run 
 the program. Works great on my thumb drive also.

 On 9/7/06, Ferguson, Michael [EMAIL PROTECTED] wrote:

   Bruce,

   How do you go about accomplishing configuring the phone, zipping it up 
 and sending it over to your family?

   Thanks

   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce 
 Reeves
   Sent: Thursday, September 07, 2006 8:37 AM

   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] Softphones IAX vs. SIP, remote 
 connectivity.

   Nick,

   I have done what you are talking about as far as being a provider for 
 family members. I used an IAX softphone mainly to eliminate the need for so 
 many holes in the firewall. And secondly because the idefisk IAX softphone 
 allowed me to extract the zip version, configure the phone, and zip the 
 folder up and email it to my family members. So for my mom it was simply 
 unzip the folder and

   On 9/7/06, Nick Ellson [EMAIL PROTECTED]  wrote:

   Bob,

   I will up the logs today, have my phone at work with me. 
 (though the Wife
   and Kids are not up yet ;)

   Anything specific I should target?

   Nick

   --
   Nick Ellson
   CCDA, CCNP, CCSP, CCAI,
   MCSE 2000, Security+, Network+
   Network Hobbyist, VFR Private Pilot.

   On Thu, 7 Sep 2006, Bob Chiodini wrote:

Nick,

Anything helpful in the asterisk or system logs.

Try bumping up the debug and verbose levels see what shows up 
 on the
console.

Weird that it would work inbound and not outbound.

Bob...

On Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote:

Hey all,

A previous annoyance with not being able to call out to my 
 brother on FWD
from my Asterisk system had me thinking that since I have my 
 own PBX, and
that system has it's own 1-to-1 static NAT to the internet, 
 I should be
able to act as the provider for him or any of my family, and 
 have them as
local extensions of my PBX, right?

So I took my laptop to work (using the X-Lite SIP softphone) 
 and watch my
ACL logs on my router for any denies to my Asterisk box. As 
 expected
udp/5060, then once that was open, a series of randomish 
 udp/1+
requests. My phone registered, and I tried to call one of 
 the phones
behind a PAP2. Worked first shot, and just as clear and 
 responsive as it
was when I was home. But, the phones at home could not call 
 me, they when
to voice mail.

I had heard that SIP doesn't survive NAT all that well, and 
 that IAX
native phones do a better job. My question is, given my 
 description of how
I am set up

RE: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.




Does anyone know off hand which IAX softphone has IM 
capabilities like XTEN?

Thanks


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Blake 
KroneSent: Thursday, September 07, 2006 3:34 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[asterisk-users] Softphones IAX vs. SIP, remote 
connectivity.
Which one has video for the mac?
On 9/7/06, Nick 
Ellson [EMAIL PROTECTED] wrote:
Hello 
  Michael,I just had both Mom and my brother up as extensions on my 
  Asterisk pbxusing IAX2, the Cubix phone for now, but I downloaded and 
  tried several. Iloke multiple lines, but a clean GUI is better for my 
  family.. Oh yeah, it worked flawlessly :)I open one port to my 
  server udp/4569 and that was it. I shut the restoff.For remote 
  family, IAX2 will be what I use right now.Anybody see a Video capable 
  version for Windows? The MAC has one, darn it. 
  Nick--Nick EllsonCCDA, CCNP, CCSP, 
  CCAI,MCSE 2000, Security+, Network+Network Hobbyist, VFR Private 
  Pilot.On Thu, 7 Sep 2006, Ferguson, Michael wrote: Hi 
  "Guys"  I too am trying to do exactly the same thing in being 
  a provider for family members. My Asterisk server is on a public ip, my home 
  is behind a Watchguard Firebox, my job is also behind a Firebox. I am using a 
  combination of Cisco 7960, Linksys 941 and XTEN Softphone. Sometimes it works 
  and sometimes it does not.  You idea on using a IAX2 softphone 
  appears to be what will solve my problem. Thanks very much 
  Post more ideas. 'preciate it. 
  -Original Message-  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] 
  ] On Behalf Of Nick Ellson Sent: Thursday, September 07, 2006 9:07 
  AM To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Subject: Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity. 
   Bruce, I *just* tested the 
  XtremePhone, IAX2 softphone. Other than trying to figure out how to get it to 
  send proper CallerID to the other phones, it worked right off, in both 
  directions. Excellent!  Perhaps working the IAX2 angle will be 
  less of a hassle, I will go looking for one that does video 
  now. Maybe it's time to buy an IAX2-ATA adaptor and see how 
  well that works over the net.  Nick As for the 
  SIP logs, I start Asterisk with -c already, I did a sip debug and tried my 
  call from the house to my remote SIP phone. YIKES!! Gunna take a bit 
  to understand all that, but I think I did see an INVITE, and a CANCEL twice in 
  a row and I did not hit the hang-up switch. So that might explain why no 
  connection is made, and the called gets my voice-mail (according to my wife) 
   -- Nick Ellson CCDA, CCNP, 
  CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, 
  VFR Private Pilot. On Thu, 7 Sep 2006, Bruce Reeves 
  wrote:  Nick, I have done what you 
  are talking about as far as being a provider for family members. I 
  used an IAX softphone mainly to eliminate the need for so many 
  holes in the firewall. And secondly because the idefisk IAX softphone 
   allowed me to extract the zip version, configure the phone, and 
  zip the folder up and email it to my family members. So for my mom 
  it was simply unzip the folder and On 
  9/7/06, Nick Ellson  [EMAIL PROTECTED] 
  wrote:Bob,I 
  will up the logs today, have my phone at work with me. (though the Wife 
  and Kids are not up yet 
  ;)Anything specific I should 
  target?Nick--Nick 
  EllsonCCDA, CCNP, CCSP, 
  CCAI,MCSE 2000, Security+, 
  Network+Network Hobbyist, VFR Private 
  Pilot.On Thu, 7 
  Sep 2006, Bob Chiodini wrote: 
  Nick,Anything 
  helpful in the asterisk or system 
  logs.Try bumping up the 
  debug and verbose levels see what shows up on the 
  console.Weird 
  that it would work inbound and not 
  outbound.Bob...On 
  Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote: 
  Hey 
  all,A previous 
  annoyance with not being able to call out to my brother 
  onFWDfrom my 
  Asterisk system had me thinking that since I have my own PBX, 
  andthat system 
  has it's own 1-to-1 static NAT to the internet, I 
  shouldbeable 
  to act as the provider for him or any of my family, and have them 
  aslocal 
  extensions of my PBX, 
  right?So I took my 
  laptop to work (using the X-Lite SIP softphone) and 
  watchmyACL 
  logs on my router for any denies to my Asterisk box. As 
  expectedudp/5060, then once that was open, 
  a series of randomish udp/1+requests. 
  My phone registered, and I tried to call one of the phones 
  behind a PAP2. Worked first shot, and just 
  as clear and responsive 
  asitwas when I 
  was home. But, the phones at home could not call me, 
  theywhen to 
  voice mail.I had 
  heard that SIP doesn't survive NAT all that well, and that 
  IAXnative phones do a better job. My 
  question is, given my description of 
  howI am set up 
  and what I am trying to accomplish, should I be looking 
  atSIPor is IAX 
  a more robust choice? (I was hoping to get vid

[asterisk-users] What are my logs telling me here?

2006-08-25 Thread Ferguson, Michael




G'Day 
All,

I am trying to 
figure out and correct some of the issues showing up in the messages log but, I 
am still a newbie and thus, somewhat at a loss, so here 
goes:

NUMBER 1 -- 
This 
appears continuously in the log REACHABLE and the 
UNREACHABLE:

Aug 
25 15:24:18 NOTICE[1867]: Peer '5103' is now REACHABLE! (418ms / 1000ms)Aug 
25 15:24:21 NOTICE[1867]: Peer '5107' is now REACHABLE! (448ms / 1000ms)Aug 
25 15:24:23 NOTICE[1867]: Peer '5108' is now REACHABLE! (445ms / 1000ms)Aug 
25 15:25:22 NOTICE[1867]: Peer '5103' is now UNREACHABLE! Last qualify: 418Aug 25 15:25:25 
NOTICE[1867]: Peer '5107' is now UNREACHABLE! Last qualify: 448Aug 25 15:25:27 
NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 445Aug 25 15:26:14 
NOTICE[1867]: Peer '5103' is now REACHABLE! (448ms / 1000ms)Aug 25 15:26:17 
NOTICE[1867]: Peer '5107' is now REACHABLE! (449ms / 1000ms)Aug 25 15:26:19 
NOTICE[1867]: Peer '5108' is now REACHABLE! (472ms / 1000ms)Aug 25 15:27:18 
NOTICE[1867]: Peer '5103' is now UNREACHABLE! Last qualify: 448Aug 25 15:27:21 
NOTICE[1867]: Peer '5107' is now UNREACHABLE! Last qualify: 449Aug 25 15:27:23 
NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 
472
NUMBER 2 -- Why 
"cause 3" and "Still have a call"

Aug 
25 11:08:47 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 460Aug 25 11:08:51 
NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3)Aug 25 
11:08:55 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3)Aug 
25 11:09:20 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 
3)Aug 25 11:09:28 NOTICE[1867]: Unable to create channel of type 'SIP' 
(cause 3)Aug 25 11:09:34 NOTICE[1867]: Peer '5103' is now REACHABLE! (457ms 
/ 1000ms)Aug 25 11:09:39 NOTICE[1867]: Peer '5108' is now REACHABLE! (551ms 
/ 1000ms)Aug 25 11:09:56 NOTICE[1867]: Still have a call...Aug 25 
11:09:56 NOTICE[1867]: Peer '5001' is now REACHABLE! (83ms / 1000ms)Aug 25 
11:40:29 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3)Aug 
25 11:40:30 NOTICE[1867]: Peer '5107' is now REACHABLE! (350ms / 1000ms)Aug 
25 11:40:31 NOTICE[1867]: Peer '5108' is now REACHABLE! (355ms / 1000ms)Aug 
25 11:49:18 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 449Aug 25 11:49:42 
NOTICE[1867]: Still have a call...Aug 25 11:49:42 NOTICE[1867]: Peer '5003' 
is now REACHABLE! (26ms / 1000ms)Aug 25 11:57:04 NOTICE[1867]: Unable to 
create channel of type 'SIP' (cause 3)Aug 25 11:58:14 NOTICE[1867]: Unable 
to create channel of type 'SIP' (cause 3)
NUMBER 3 -- This is 
also repeated quite a bit.

Aug 
24 14:36:30 WARNING[8809]: Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Non-critical Request)Aug 24 14:36:30 WARNING[8809]: Maximum 
retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Non-critical Request)Aug 24 14:36:30 WARNING[8809]: Maximum 
retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Non-critical Request)
Any pointers, 
documents, help criticisms welcome..Thanks...Mike



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RE: [asterisk-users] What are my logs telling me here?

2006-08-25 Thread Ferguson, Michael

 BJ,
Thanks much. I do have qualify in my sip.conf (see below) set at 1000.
Also, the asterisk box sits on a public ip ( no firewall) but the devices are 
behind a WatchGuard firewall.

Thanks for the pointers. Send me more if you have any. Thanks


[5002]
type=friend ; either friend (peer+user), peer or user
host=dynamic
username=5002
secret=5002
context=toll-access
canreinvite=no
qualify=1000
callerid=5002
disallow=all
allow=ulaw
allow=alaw
[EMAIL PROTECTED]
nat=yes
dtmfmode=rfc2833

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Friday, August 25, 2006 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What are my logs telling me here?

On 8/25/06, Ferguson, Michael [EMAIL PROTECTED] wrote:


 G'Day All,

 I am trying to figure out and correct some of the issues showing up in 
 the messages log but, I am still a newbie and thus, somewhat at a 
 loss, so here
 goes:

 NUMBER 1 -- This appears continuously in the log REACHABLE and the
 UNREACHABLE:


 Aug 25 15:24:18 NOTICE[1867]: Peer '5103' is now REACHABLE! (418ms / 
 1000ms) Aug 25 15:24:21 NOTICE[1867]: Peer '5107' is now REACHABLE! 
 (448ms / 1000ms) Aug 25 15:24:23 NOTICE[1867]: Peer '5108' is now 
 REACHABLE! (445ms / 1000ms) Aug 25 15:25:22 NOTICE[1867]: Peer '5103' is now 
 UNREACHABLE!  Last qualify:
 418
 Aug 25 15:25:25 NOTICE[1867]: Peer '5107' is now UNREACHABLE!  Last qualify:
 448
 Aug 25 15:25:27 NOTICE[1867]: Peer '5108' is now UNREACHABLE!  Last qualify:
 445
 Aug 25 15:26:14 NOTICE[1867]: Peer '5103' is now REACHABLE! (448ms / 
 1000ms) Aug 25 15:26:17 NOTICE[1867]: Peer '5107' is now REACHABLE! 
 (449ms / 1000ms) Aug 25 15:26:19 NOTICE[1867]: Peer '5108' is now 
 REACHABLE! (472ms / 1000ms) Aug 25 15:27:18 NOTICE[1867]: Peer '5103' is now 
 UNREACHABLE!  Last qualify:
 448
 Aug 25 15:27:21 NOTICE[1867]: Peer '5107' is now UNREACHABLE!  Last qualify:
 449
 Aug 25 15:27:23 NOTICE[1867]: Peer '5108' is now UNREACHABLE!  Last qualify:
 472

 NUMBER 2 -- Why cause 3 and Still have a call


 Aug 25 11:08:47 NOTICE[1867]: Peer '5108' is now UNREACHABLE!  Last qualify:
 460
 Aug 25 11:08:51 NOTICE[1867]: Unable to create channel of type 'SIP' 
 (cause
 3)
 Aug 25 11:08:55 NOTICE[1867]: Unable to create channel of type 'SIP' 
 (cause
 3)
 Aug 25 11:09:20 NOTICE[1867]: Unable to create channel of type 'SIP' 
 (cause
 3)
 Aug 25 11:09:28 NOTICE[1867]: Unable to create channel of type 'SIP' 
 (cause
 3)
 Aug 25 11:09:34 NOTICE[1867]: Peer '5103' is now REACHABLE! (457ms / 
 1000ms) Aug 25 11:09:39 NOTICE[1867]: Peer '5108' is now REACHABLE! 
 (551ms / 1000ms) Aug 25 11:09:56 NOTICE[1867]: Still have a call...
 Aug 25 11:09:56 NOTICE[1867]: Peer '5001' is now REACHABLE! (83ms / 
 1000ms) Aug 25 11:40:29 NOTICE[1867]: Unable to create channel of type 
 'SIP' (cause
 3)
 Aug 25 11:40:30 NOTICE[1867]: Peer '5107' is now REACHABLE! (350ms / 
 1000ms) Aug 25 11:40:31 NOTICE[1867]: Peer '5108' is now REACHABLE! 
 (355ms / 1000ms) Aug 25 11:49:18 NOTICE[1867]: Peer '5108' is now 
 UNREACHABLE!  Last qualify:
 449
 Aug 25 11:49:42 NOTICE[1867]: Still have a call...
 Aug 25 11:49:42 NOTICE[1867]: Peer '5003' is now REACHABLE! (26ms / 
 1000ms) Aug 25 11:57:04 NOTICE[1867]: Unable to create channel of type 
 'SIP' (cause
 3)
 Aug 25 11:58:14 NOTICE[1867]: Unable to create channel of type 'SIP' 
 (cause
 3)

 NUMBER 3 -- This is also repeated quite a bit.


 Aug 24 14:36:30 WARNING[8809]: Maximum retries exceeded on call
 [EMAIL PROTECTED] for seqno 102 
 (Non-critical Request) Aug 24 14:36:30 WARNING[8809]: Maximum retries 
 exceeded on call
 [EMAIL PROTECTED] for seqno 102 
 (Non-critical Request) Aug 24 14:36:30 WARNING[8809]: Maximum retries 
 exceeded on call
 [EMAIL PROTECTED] for seqno 102 
 (Non-critical Request)

 Any pointers, documents, help criticisms 
 welcome..Thanks...Mike


 You've probably got qualify= on your peers in sip.conf. So Asterisk is sending 
out a SIP OPTIONS msg to which it's waiting for the peer's reply. If it doesn't 
respond, it then marks the peer as unreachable, and you then cannot dial out to 
the peer because it's state is UNREACHABLE which will cause (status 3) messages.

 You might consider increasing your qualify= time and see if that corrects your 
problems. If not, you're going to need to start looking at possible 
firewall/network interruptions between your Asterisk instance and your devices 
to see if they are knocking down traffic that might be trying to flow between.

 BJ

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[asterisk-users] Configure mailserver to deliver voicemail

2006-08-21 Thread Ferguson, Michael






G'Day 
List,

I am looking for 
documentation on how to configure sendmail to deliver asterisk voicemails to the 
recipient's mailbox.
I Googled it but 
found many many references to the fact that asterisk can do that but no 
How-To's.

I believe sendmail 
is running on my asterisk box as:
[EMAIL PROTECTED] /] # mail 

returns... Mail 
version 8.1 6/6/93

Also, my 
voicemail.conf is already configured.

Thanks
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RE: [asterisk-users] Registration Error

2006-08-18 Thread Ferguson, Michael

Olle,

Thanks
,preciate it.

Best Wishes 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, August 18, 2006 3:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Registration Error


17 aug 2006 kl. 18.38 skrev Ferguson, Michael:


 G'Day List;

 I hoping for some direction here:

 The following message is scrolling without end on my asterisk box,
 continuously: (NOTE: date and time changes accordingly and IP 
 addresses are not real)

 Aug 17 11:49:53 NOTICE[1034]: chan_sip.c:8038 handle_request:  
 Registration from 'sip:[EMAIL PROTECTED];user=phone' Failed for 
 '64.64.64.12'
 Aug 17 11:49:53 NOTICE[1034]: chan_sip.c:8038 handle_request:  
 Registration from 'sip:[EMAIL PROTECTED];user=phone' Failed for 
 '64.64.12.12'
 Aug 17 11:49:53 NOTICE[1034]: chan_sip.c:8038 handle_request:  
 Registration from 'sip:[EMAIL PROTECTED];user=phone' Failed for 
 '64.64.12.12'
 Aug 17 11:49:53 NOTICE[1034]: chan_sip.c:8038 handle_request:  
 Registration from 'sip:[EMAIL PROTECTED];user=phone' Failed for 
 '64.64.12.12'

 Just so you know, the asterisk box sits on a public IP
 (64.64.64.64) that's on the same subnet as my firewall (64.64.64.12), 
 behind which, my 7960 sits.

 Any thoughts?

I would suggest reading the message. The device can't register with Asterisk - 
propably an authentication error.
Check the password for account 5104 both in the phone and in sip.conf and make 
sure they are the same.

/Olle

---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk beachcamp: Bootcamp in Malaga, Spain - http://edvina.net/



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[asterisk-users] Registration Error

2006-08-17 Thread Ferguson, Michael





G'Day 
List;

I hoping for some 
direction here:



The following messageisscrolling 
without end on my asterisk box, continuously: 
(NOTE: date and time changes accordingly and IP addresses are not 
real)

Aug 17 11:49:53 
NOTICE[1034]: chan_sip.c:8038 handle_request: Registration from 
'sip:[EMAIL PROTECTED];user=phone' Failed for '64.64.64.12'

Aug 17 11:49:53 
NOTICE[1034]: chan_sip.c:8038 handle_request: Registration from 
'sip:[EMAIL PROTECTED];user=phone' Failed for 
'64.64.12.12'

Aug 17 11:49:53 
NOTICE[1034]: chan_sip.c:8038 handle_request: Registration from 
'sip:[EMAIL PROTECTED];user=phone' Failed for 
'64.64.12.12'

Aug 17 11:49:53 
NOTICE[1034]: chan_sip.c:8038 handle_request: Registration from 
'sip:[EMAIL PROTECTED];user=phone' Failed for 
'64.64.12.12'


Just so you know, the asterisk box sits on a public IP 
(64.64.64.64) that's on the same subnet as my firewall(64.64.64.12), behind 
which, my 7960 sits.

Any 
thoughts?

Thanks

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[asterisk-users] Frustration cubed

2006-08-17 Thread Ferguson, Michael




Hello 
All,

I am quite frustrated at my 
lack of knowledge here and so I seek pointers from you, the wise 
ones.
Repeated scouring of my 
.conf files is unfruitfull.

Problem 
#1
From across the WAN both 
phones connect to my asterisk box but, while the Grandstream101(ext 
5001)can call the Cisco 7960(ext5103) and have conversation, when the 7960 
calls the Grandstream, there is no ring but an immediate reply saying that the 
person at extension 5001 is on the phone. Please leave a 
message.

I can't find the source of 
the problem. Any pointers as to where to look?


Problem 
#2

The following messageisscrolling 
without end on my asterisk box, continuously: 
(NOTE: date and time changes accordingly and IP addresses are not 
real)

Aug 17 11:49:53 
NOTICE[1034]: chan_sip.c:8038 handle_request: Registration from 
'sip:[EMAIL PROTECTED];user=phone' Failed for '64.64.64.12'

Aug 17 11:49:53 
NOTICE[1034]: chan_sip.c:8038 handle_request: Registration from 
'sip:[EMAIL PROTECTED];user=phone' Failed for 
'64.64.12.12'

Any pointers here?
--

Problem#3
After configuing/updating the Cisco 7960 on my office LAN with the 
TFTP Server, I took the 7960 home. Now the additional 
RingTones
no longer appear. So, am I to conclude that the RingTones do not 
remain on the 7960 if it cannot find the TFTP server?


Any pointers/suggestions welcome.

Thanks.

Ferg



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RE: [asterisk-users] Cisco 7960 password reset

2006-08-16 Thread Ferguson, Michael

David and Barry,

Thanks for the help.

'preciate it. 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop
Sent: Wednesday, August 16, 2006 6:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960 password reset

If the phone already had the SIP image running.
Check the SIPDefault.cnf file there may be a phone_password= string this is 
the phone's current password use it remember to change to number or uppercase 
if need be

Ferguson, Michael wrote:
 Maxx,

 Thanks much for the feedback. I will check into it and follow up with
 your instructions.

 'preciate it. Best wishes.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo
 Sent: Tuesday, August 15, 2006 5:48 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Cisco 7960 password reset

 What Cisco image is the phone running? If it is really old (lower than
 P0S030203) then yeah, this won't work.

 If you upgrade the phone to P0S30203 and from there on to P0S3-06-3-00,
 and then these instructions will work fine. This should be pretty
 straightforward using ATFTP and the Cisco images.

 In response to your other question, a factory reset TMK does not wipe
 out the SIP image. Just the settings.

 --Maxx

 Ferguson, Michael wrote:

 Maxx,
 That did not work.
 Any other ideas?

 Thanks

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Maxx 
 Lobo
 Sent: Tuesday, August 15, 2006 4:43 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Cisco 7960 password reset

 Fastest way (wipes everything out):

 1. Power off the phone completely.
 2. Hold down the # key, then power the phone on.
 3. Continue holding the # key until the LCD gives you a status

 message.

 4. Follow the prompts to do a full factory reset, which resets the

 password as well.

 --Maxx

 Ferguson, Michael wrote:

 G'Day List,

 I am trying, once again, to configure my 7960 to work with asterisk.
 Where abouts do I go to reset the password on the phone?

 Thanks

 -
 -
 --

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[asterisk-users] Cisco 7960 password reset




G'Day 
List,

I am trying, once 
again, to configure my 7960 to work with asterisk.
Where abouts do I go 
to reset the password on the phone?

Thanks



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RE: [asterisk-users] Cisco 7960 password reset

Thanks.

Will this action blow away the SIP images I already have on the phone?

'preciate it. 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo
Sent: Tuesday, August 15, 2006 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960 password reset

Fastest way (wipes everything out):

1. Power off the phone completely.
2. Hold down the # key, then power the phone on.
3. Continue holding the # key until the LCD gives you a status message.
4. Follow the prompts to do a full factory reset, which resets the password as 
well.

--Maxx

Ferguson, Michael wrote:
 G'Day List,

 I am trying, once again, to configure my 7960 to work with asterisk.
 Where abouts do I go to reset the password on the phone?

 Thanks

 --
 --

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RE: [asterisk-users] Cisco 7960 password reset

Maxx,
That did not work.
Any other ideas?

Thanks 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo
Sent: Tuesday, August 15, 2006 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960 password reset

Fastest way (wipes everything out):

1. Power off the phone completely.
2. Hold down the # key, then power the phone on.
3. Continue holding the # key until the LCD gives you a status message.
4. Follow the prompts to do a full factory reset, which resets the password as 
well.

--Maxx

Ferguson, Michael wrote:
 G'Day List,

 I am trying, once again, to configure my 7960 to work with asterisk.
 Where abouts do I go to reset the password on the phone?

 Thanks

 --
 --

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RE: [asterisk-users] Cisco 7960 password reset

Maxx,

Thanks much for the feedback. I will check into it and follow up with
your instructions.

'preciate it. Best wishes.










 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo
Sent: Tuesday, August 15, 2006 5:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960 password reset

What Cisco image is the phone running? If it is really old (lower than
P0S030203) then yeah, this won't work.

If you upgrade the phone to P0S30203 and from there on to P0S3-06-3-00,
and then these instructions will work fine. This should be pretty
straightforward using ATFTP and the Cisco images.

In response to your other question, a factory reset TMK does not wipe
out the SIP image. Just the settings.

--Maxx

Ferguson, Michael wrote:
 Maxx,
 That did not work.
 Any other ideas?
 
 Thanks
 
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Maxx 
 Lobo
 Sent: Tuesday, August 15, 2006 4:43 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Cisco 7960 password reset
 
 Fastest way (wipes everything out):
 
 1. Power off the phone completely.
 2. Hold down the # key, then power the phone on.
 3. Continue holding the # key until the LCD gives you a status
message.
 4. Follow the prompts to do a full factory reset, which resets the
password as well.
 
 --Maxx
 
 Ferguson, Michael wrote:
 G'Day List,
  
 I am trying, once again, to configure my 7960 to work with asterisk.
 Where abouts do I go to reset the password on the phone?
  
 Thanks
  
  


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[asterisk-users] Steve Totaro I am trying to reach you.

2006-08-04 Thread Ferguson, Michael




Where can you be 
found?

Ferguson
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[Asterisk-Users] 7960 Not Picking up new firmware.

2005-02-23 Thread Ferguson, Michael

G'Day All.

So I got the TFTP server all set up -thanks to much help from this list-
the 7960 found it and updated to SIP the first firmware P0S30200. What I
am now trying to do is upgrate through all the versions, as recommended,
to the latest version, P003-07-3-00.

I thought this would be accomplished by simply changing the sole line in
the OS79XX.TXT file to P0S30203 and reboot the phone. But no success.

Any pointers? Thanks


Ferg

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RE: [Asterisk-Users] 7960 Not Picking up new firmware.

2005-02-23 Thread Ferguson, Michael

Gary,
Thanks again. You help has been invaluable. 'preciate it

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary G.
Hendershot
Sent: Wednesday, February 23, 2005 11:26 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] 7960 Not Picking up new firmware.

You have to change the image name in the OS79XX.txt and SIPDefault.cnf
files to match the name of BIN file you are trying to load ... With
versions of the firmware prior to 7.x, the name you put in the
OS79XX.txt file and the SIPDefault.cnf files are the same; simply the
BIN file name less the BIN extension  ...  As you get to version 7.x and
up, the file name you put in OS79XX.txt is actually the name of a
Universal Loader ... The name of the SIP binary image is entered in
SIPDefault.cnf ...

I got a help on this one from a pretty decent article on the WIKI at
http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx  ...  Look at
the section header Software Upgrade Requirements ...  This gave me the
clues I needed to get the 7.3 Sip image to load properly ...

G.Hendershot

-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, February 23, 2005 10:43 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] 7960 Not Picking up new firmware.

G'Day All.

So I got the TFTP server all set up -thanks to much help from this list-
the 7960 found it and updated to SIP the first firmware P0S30200. What I
am now trying to do is upgrate through all the versions, as recommended,
to the latest version, P003-07-3-00.

I thought this would be accomplished by simply changing the sole line in
the OS79XX.TXT file to P0S30203 and reboot the phone. But no success.

Any pointers? Thanks

Ferg

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[Asterisk-Users] TFTP Server

G'Day All,

Can anyone give me some direction in setting up the TFTP server on my
RadHat ES3 box?

I did quite a bit of reading, but I think I am more unsure now than
before. I found the information nebulous. TFTP is already installed. I
am trying to determine where the root directory for the tftp services is
located so I can copy the CISCO 7960 firmware files onto it.
 Thanks Ferg
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RE: [Asterisk-Users] TFTP Server

Thanks Clay Reiche.

Anyone,
Why is the 7960 looking for a call manager at 168.254.173.1?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Clay
Reiche
Sent: Tuesday, February 22, 2005 1:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TFTP Server


Edit /etc/xinetd.d/tftp
The -s argument is the root directory and make sure you set disable =
no.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Tuesday, February 22, 2005 1:25 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] TFTP Server

G'Day All,

Can anyone give me some direction in setting up the TFTP server on my
RadHat ES3 box?

I did quite a bit of reading, but I think I am more unsure now than
before. I found the information nebulous. TFTP is already installed. I
am trying to determine where the root directory for the tftp services is
located so I can copy the CISCO 7960 firmware files onto it.  Thanks
Ferg ___
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RE: [Asterisk-Users] TFTP Server

I created a different dir, /SIPFONE
Now I have to check if it readable by all. Thanks.

I set my Windows 2003 DHCP to assign the TFTP server's IP address,
default gateway, dns, etc, etc and the phone got all that quite well but
not picking up the files.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rudolf
Ladyzhenskii
Sent: Tuesday, February 22, 2005 5:25 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] TFTP Server


Hi,

setup is in /etc/xinet.d/tftp file

Default directory is /tftpboot. make sure that this directory is
readable by anyone.

Rudolf



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Gary G.
Hendershot
Sent: Wednesday, February 23, 2005 9:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] TFTP Server


On my server (ES3) the TFTPBOOT folder is where I put my Cisco image
loader files 

-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, February 22, 2005 1:25 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] TFTP Server

G'Day All,

Can anyone give me some direction in setting up the TFTP server on my
RadHat ES3 box?

I did quite a bit of reading, but I think I am more unsure now than
before. I found the information nebulous. TFTP is already installed. I
am trying to determine where the root directory for the tftp services is
located so I can copy the CISCO 7960 firmware files onto it.  Thanks
Ferg


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RE: [Asterisk-Users] TFTP Server

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rudolf
Ladyzhenskii
Sent: Tuesday, February 22, 2005 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TFTP Server


Any directory name is fine as long as you configured TFTP server to use
it.

Also, from device (phone) point of view, your /TFTPBOOT directory is '/'
(root) directory on server!

Rudolf

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ferguson,
Michael
Sent: Wednesday, February 23, 2005 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TFTP Server


I created a different dir, /SIPFONE
Now I have to check if it readable by all. Thanks.

I set my Windows 2003 DHCP to assign the TFTP server's IP address,
default gateway, dns, etc, etc and the phone got all that quite well but
not picking up the files.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rudolf
Ladyzhenskii
Sent: Tuesday, February 22, 2005 5:25 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] TFTP Server


Hi,

setup is in /etc/xinet.d/tftp file

Default directory is /tftpboot. make sure that this directory is
readable by anyone.

Rudolf



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Gary G.
Hendershot
Sent: Wednesday, February 23, 2005 9:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] TFTP Server


On my server (ES3) the TFTPBOOT folder is where I put my Cisco image
loader files 

-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, February 22, 2005 1:25 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] TFTP Server

G'Day All,

Can anyone give me some direction in setting up the TFTP server on my
RadHat ES3 box?

I did quite a bit of reading, but I think I am more unsure now than
before. I found the information nebulous. TFTP is already installed. I
am trying to determine where the root directory for the tftp services is
located so I can copy the CISCO 7960 firmware files onto it.  Thanks
Ferg


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RE: [Asterisk-Users] This is NUTS!!SOLVED

2005-02-20 Thread Ferguson, Michael

Title: Message



So 
true.

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Ed 
  BradySent: Saturday, February 19, 2005 10:50 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] This is NUTS!!SOLVEDMakes you wonder 
  about the future of CISCO doen't it?  You are a potential customer 
  trying every means possible to give them money, and they are making it 
  difficult to do so. Most thriving businesses usually make it as 
  convenient as possible for their customers to give them money. This 
  reminds me of similar stories of Digital Equipment Corporation (DEC) 
  before they fell on hard times.Ferguson, Michael wrote: 
  Thanks everyone for your feedback, especially Mark. I now have the ALL
the files I need. My order still stands for the $8.00 product from CISCO
but the CP7960 dealer sent me all the files.

Now I will move on to completeing the setup of the TFTP server. Thanks
again


  

-Original Message-
From: Michael Loftis [mailto:[EMAIL PROTECTED]] 
Sent: Friday, February 18, 2005 7:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Ferguson, Michael
Subject: Re: [Asterisk-Users] This is NUTS!!




--On Friday, February 18, 2005 10:21 -0500 "Ferguson, Michael" 
[EMAIL PROTECTED] wrote:

  
G'Day All;

So I purchased a Cisco 7960 and am now trying to get it configured for

  
*. No can do without the variuos files/images through a FTPF server. I
configured the TFTP server on my RHES 3 box, now to get the required
CISCO files.

So I contacted CISCO to purchase the required maintenance contract so 
as to gain access to the download area for the files/images. -WHAT A
FRUSTRATION!!-

CISCO says, "Purchase it from your reseller/dealer."  OK. So I call my

  
reseller/dealer and he is having the most difficult time getting this 
$8.00 product, CON-SNT-CP7960, for me. It is just not worth the time 
and effort for him. So here I am, a week later, and no CP7960. It 
looks pretty though!!

Can anyone recommend a speedier way to get this CON-SNT-CP7960 from 
CISCO

Try contacting CDW, you'll need the phones serial number but they can 
probably help you out and get you the SMARTnet package.

--
GPG/PGP -- 0xE736BD7E 5144 6A2D 977A 6651 DFBE 1462 E351 88B9 E736 BD7E

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RE: [Asterisk-Users] This is NUTS!!SOLVED

2005-02-19 Thread Ferguson, Michael

Thanks everyone for your feedback, especially Mark. I now have the ALL
the files I need. My order still stands for the $8.00 product from CISCO
but the CP7960 dealer sent me all the files.

Now I will move on to completeing the setup of the TFTP server. Thanks
again


  

-Original Message-
From: Michael Loftis [mailto:[EMAIL PROTECTED] 
Sent: Friday, February 18, 2005 7:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Ferguson, Michael
Subject: Re: [Asterisk-Users] This is NUTS!!




--On Friday, February 18, 2005 10:21 -0500 Ferguson, Michael 
[EMAIL PROTECTED] wrote:

 G'Day All;

 So I purchased a Cisco 7960 and am now trying to get it configured for

 *. No can do without the variuos files/images through a FTPF server. I
 configured the TFTP server on my RHES 3 box, now to get the required
 CISCO files.

 So I contacted CISCO to purchase the required maintenance contract so 
 as to gain access to the download area for the files/images. -WHAT A
 FRUSTRATION!!-

 CISCO says, Purchase it from your reseller/dealer.  OK. So I call my

 reseller/dealer and he is having the most difficult time getting this 
 $8.00 product, CON-SNT-CP7960, for me. It is just not worth the time 
 and effort for him. So here I am, a week later, and no CP7960. It 
 looks pretty though!!

 Can anyone recommend a speedier way to get this CON-SNT-CP7960 from 
 CISCO

Try contacting CDW, you'll need the phones serial number but they can 
probably help you out and get you the SMARTnet package.

--
GPG/PGP -- 0xE736BD7E 5144 6A2D 977A 6651 DFBE 1462 E351 88B9 E736 BD7E

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[Asterisk-Users] This is NUTS!!

2005-02-18 Thread Ferguson, Michael

G'Day All;

So I purchased a Cisco 7960 and am now trying to get it configured for
*.
No can do without the variuos files/images through a FTPF server. I
configured the TFTP server on my RHES 3 box, now to get the required
CISCO files.

So I contacted CISCO to purchase the required maintenance contract so as
to gain access to the download area for the files/images. -WHAT A
FRUSTRATION!!-

CISCO says, Purchase it from your reseller/dealer.  OK. So I call my
reseller/dealer and he is having the most difficult time getting this
$8.00 product, CON-SNT-CP7960, for me. It is just not worth the time and
effort for him. So here I am, a week later, and no CP7960. It looks
pretty though!!

Can anyone recommend a speedier way to get this CON-SNT-CP7960 from
CISCO

Thanks

Ferg
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RE: [Asterisk-Users] Re: Cisco 7970 Won't boot after factory reset

2005-02-17 Thread Ferguson, Michael

Colin, Thanks for that pointer. 'preciate it

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Thursday, February 17, 2005 12:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Cisco 7970 Won't boot after factory
reset

how does the phone know where to find the TFTP server..?

Dude, option 150 in your DHCP server:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186
a00800942f4.shtml

We use the same option for our Mitel phones. HTH. 
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[Asterisk-Users] TFTP Serer ????

2005-02-14 Thread Ferguson, Michael

G'Day All,
Can someone help me out please. My new CISCO 7960's manual says I have
to setup a TFTP server. Googled it and got a little understanding, but
from * standpoint, well I am still a lost.
Can I set this tftp server on the same * box? Can in be on a WinXP box?
Which tftp software would you recommend?

Thanks much.

BTY: Does anyone have a How-To on getting the 7960 fully configured for
*?

ThanksFerg
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RE: [Asterisk-Users] TFTP Serer ????

2005-02-14 Thread Ferguson, Michael

Stefan,
Thanks a million. 'preciate it.




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stefan
Gofferje
Sent: Monday, February 14, 2005 8:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TFTP Serer 


Ferguson, Michael schrieb:
 G'Day All,
 Can someone help me out please. My new CISCO 7960's manual says I have

 to setup a TFTP server. Googled it and got a little understanding, but

 from * standpoint, well I am still a lost. Can I set this tftp server 
 on the same * box? Can in be on a WinXP box? Which tftp software would

 you recommend?

Any Linux distro should ship with one or two tftp servers. Anyway, away 
from firmware updates, the config could be done via phone menu or 
webinterface. There also are various tftpds available for Windows.

 BTY: Does anyone have a How-To on getting the 7960 fully configured 
 for *?

http://www.voip-info.org/tiki-index.php?page=cisco%2079xx
http://www.voip-info.org/tiki-index.php?page=Setup%20SiP%20on%207940%20-
%207960

Regards,
   Stefan

-- 
  (o_   Stefan Gofferje  | Linux Systems Specialist
  //\   Reg'd Linux User #247167 | Network Security Specialist
  V_/_  Linux is like a Wigwam - No gates, no windows, Apache inside

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[Asterisk-Users] Quick How-To Guide for getting a Cisco 7960 going.

2005-02-11 Thread Ferguson, Michael

G'Day All,

I just received my new 7960 and, while I am still reading the manuals
from the web and the At a Glance booklet that came with the phone, I
am hoping that someone can point me to a some real quick steps to get
the phone registered with my * server and in working order. I will keep
reading the docs though.

Thanks 

Ferg
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[Asterisk-Users] Messaging with * and eyeBeam

2005-02-01 Thread Ferguson, Michael

G'Day All,

Eyebeam has gotten my interest but I do not have a high-altitude view
of its interraction with *, therefore my questions. 
I called xTEN but they preferr to talk to telcos and ISP's purchasing
hundreds of the eyebeam software... Kind-a stuck here.

I already have * happily running and taking care of business on my
Windows network. I also have Windows IM running on my exchange server
and users a IM'ing quite a bit - all internal though-.

While I would prefer to use * and eyebeam, I am quite cautious about
installing another piece of software on users workstations for them to
learn. Plus it seems that Windows messaging can do just the same as
eyeBeam.

What are your thoughts??? 


Thanks 
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[Asterisk-Users] FW: Messaging with * and eyeBeam

2005-02-01 Thread Ferguson, Michael

-Original Message-
From: Ferguson, Michael 
Sent: Tuesday, February 01, 2005 11:35 AM
To: 'asterisk-users@lists.digium.com'
Subject: Messaging with * and eyeBeam

G'Day All,

Eyebeam has gotten my interest but I do not have a high-altitude view
of its interraction with *, therefore my questions. 
I called xTEN but they preferr to talk to telcos and ISP's purchasing
hundreds of the eyebeam software... Kind-a stuck here.

I already have * happily running and taking care of business on my
Windows network. I also have Windows IM running on my exchange server
and users a IM'ing quite a bit - all internal though-.

While I would prefer to use * and eyebeam, I am quite cautious about
installing another piece of software on users workstations for them to
learn. Plus it seems that Windows messaging can do just the same as
eyeBeam.

What are your thoughts??? 

Thanks 
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[Asterisk-Users] Eyebeam Vs. Windows Messanger,

2005-01-31 Thread Ferguson, Michael

G'Day All,
Eyebeam has gotten my interest but I do not have a high-altitude view
of its interraction with *, therefore my questions.
I already have * happily running and taking care of business on my
Windows network. I also have Windows IM running on my exchange server
and users a IM'ing quite a bit - all internal though-.

While I would prefer to use * and eyebeam, I am quite cautious about
installing another piece of software on users workstations for them to
learn. Plus it seems that Windows messaging can do just the same as
eyeBeam.

What are your thoughts??? 

Thanks 





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RE: [Asterisk-Users] Eyebeam - asterisk - Messenger

2005-01-29 Thread Ferguson, Michael

I am curious about Eyebeam so I went to xten's site and read up on it.
I still do not get a clear understanding as to what Eyebeam does. Help
me to understand it:

Am I correct?
It installs on a windows computer?
It connects/registers to my * box?
With a camera attached to my Windows computer I can use Eyebeam to make
video calls on the lan or wan as long as I know the recipient's IP
address?

No?

Thanks

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ing.
Ignacio Ortega A.
Sent: Saturday, January 29, 2005 12:18 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Eyebeam - asterisk - Messenger


i did evrything you mentioned, i thing is for my eyebeam version, mine
is 3002s what`s yours?


On Fri, 28 Jan 2005 23:10:40 -0500 (EST), Francois Meehan
[EMAIL PROTECTED] wrote:
 Thanks Wessel,
 
 You really have to know about that little switch on button, I had 2 
 eyebeam connected with their cameras, no video, 5 min. after I tried 
 the little button and it worked. Must be the effect of the first rhum 
 of the week-end...
 
 Tried with (windows) messenger, it would not go. That is why I bought 
 another eyebeam and it's all working now. Thanks to the vpn, video 
 communications from the outside work like a charm.
 
 Guess I will stick with eyebeam for now.
 
 Also, for the record, in addition to videosupport=yes, the video 
 codecs must be enabled in the sip.conf:
 
 allow=h261
 allow=h263
 
 and for each phone I have put:
 
 canreinvite=no
 
 
 Have a good week-end,
 
 Francois
 
  Just add a line to your sip.conf:
  [general]
  videosupport=yes
 
 
  And to enable video with eyeBeam press the switchon button on the 
  screen
  :-)
 
  Wessel
 
  -Original Message-
  From: Ing. Ignacio Ortega A. [mailto:[EMAIL PROTECTED]
  Sent: Friday, January 28, 2005 19:33
  To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial 
  Discussion
  Subject: Re: [Asterisk-Users] Eyebeam - asterisk - Messenger
 
  did you find how to configure video with eyebeam using asterisk 
  because i wasn`t able to do it yet
 
  as well i want to se messangin with it
 
  ThanK You
 
 
  On Fri, 28 Jan 2005 13:23:46 -0500 (EST), Francois Meehan 
  [EMAIL PROTECTED] wrote:
   Hi all,
  
   I would like to connect in sip mode an Eyebeam client to a
  messenger
   via Asterisk.
  
   I want to use video.
  
   Nat is not an issue as vpn connections will be used.
  
   Is this a difficult tasks, can someone give me some pointers to 
   get started...
  
   Have a good week-end,
  
   Francois
  
   Random Thought:
   ---
   Wanna buy a duck? ___
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 ---
 When the eyes say one thing and the tongue another, a practiced man 
 relies on the language of the first. - Ralph Waldo Emerson, 1803 - 
 1882 ___
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RE: [Asterisk-Users] Eyebeam - asterisk - Messenger

2005-01-29 Thread Ferguson, Michael

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean
collins
Sent: Saturday, January 29, 2005 8:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Ing.
Ignacio Ortega A.
Subject: RE: [Asterisk-Users] Eyebeam - asterisk - Messenger


Yep, basically it is a SIP video phone like a grandstream is a SIP voice
phone.

Cheers,
Dean

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Saturday, January 29, 2005 6:44 PM
To: Ing. Ignacio Ortega A.; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Eyebeam - asterisk - Messenger

I am curious about Eyebeam so I went to xten's site and read up on it. I
still do not get a clear understanding as to what Eyebeam does. Help me
to understand it:

Am I correct?
It installs on a windows computer?
It connects/registers to my * box?
With a camera attached to my Windows computer I can use Eyebeam to make
video calls on the lan or wan as long as I know the recipient's IP
address?

No?

Thanks

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ing.
Ignacio Ortega A.
Sent: Saturday, January 29, 2005 12:18 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Eyebeam - asterisk - Messenger


i did evrything you mentioned, i thing is for my eyebeam version, mine
is 3002s what`s yours?


On Fri, 28 Jan 2005 23:10:40 -0500 (EST), Francois Meehan
[EMAIL PROTECTED] wrote:
 Thanks Wessel,
 
 You really have to know about that little switch on button, I had 2
 eyebeam connected with their cameras, no video, 5 min. after I tried 
 the little button and it worked. Must be the effect of the first rhum 
 of the week-end...
 
 Tried with (windows) messenger, it would not go. That is why I bought
 another eyebeam and it's all working now. Thanks to the vpn, video 
 communications from the outside work like a charm.
 
 Guess I will stick with eyebeam for now.
 
 Also, for the record, in addition to videosupport=yes, the video
 codecs must be enabled in the sip.conf:
 
 allow=h261
 allow=h263
 
 and for each phone I have put:
 
 canreinvite=no
 
 
 Have a good week-end,
 
 Francois
 
  Just add a line to your sip.conf:
  [general]
  videosupport=yes
 
 
  And to enable video with eyeBeam press the switchon button on the
  screen
  :-)
 
  Wessel
 
  -Original Message-
  From: Ing. Ignacio Ortega A. [mailto:[EMAIL PROTECTED]
  Sent: Friday, January 28, 2005 19:33
  To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
  Discussion
  Subject: Re: [Asterisk-Users] Eyebeam - asterisk - Messenger
 
  did you find how to configure video with eyebeam using asterisk
  because i wasn`t able to do it yet
 
  as well i want to se messangin with it
 
  ThanK You
 
 
  On Fri, 28 Jan 2005 13:23:46 -0500 (EST), Francois Meehan
  [EMAIL PROTECTED] wrote:
   Hi all,
  
   I would like to connect in sip mode an Eyebeam client to a
  messenger
   via Asterisk.
  
   I want to use video.
  
   Nat is not an issue as vpn connections will be used.
  
   Is this a difficult tasks, can someone give me some pointers to
   get started...
  
   Have a good week-end,
  
   Francois
  
   Random Thought:
   ---
   Wanna buy a duck? ___
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 When the eyes say one thing and the tongue another, a practiced man
 relies on the language of the first. - Ralph Waldo Emerson, 1803 - 
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RE: [Asterisk-Users] Webmin Module for Asterisk (and thirdlane)

2005-01-21 Thread Ferguson, Michael

Same here.
I called them yesterday plus email and still no reply.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, January 21, 2005 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Webmin Module for Asterisk (and thirdlane)

Henry Devito wrote:

 www.thirdlane.com http://www.thirdlane.com  has already written a
 close dsource webmin module.  I have no idea how much it costs or how 
 well it works.

I've attempted to contact thirdlane to get pricing on their GUI and 
can't seem to get anyone to reply.

My personal feeling is that if it's closed source, the support better be

excellent. And if I can't get a reply to a sales question.. What's going

to happen when I have a problem?

Ek!
-Brett

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[Asterisk-Users] Jamaica DID

2005-01-17 Thread Ferguson, Michael

G'Day All,

Any recommendations for getting a few DID's for Jamaica? Area code 876?

Thanks
Ferg
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[Asterisk-Users] Jamaica

2005-01-17 Thread Ferguson, Michael

G'Day,
 Can any one recommend a reliable source for DID's for Jamaica?

Thanks
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RE: [Asterisk-Users] Jamaica - My apologies for the second post.

2005-01-17 Thread Ferguson, Michael

My apologies for the second post. I thought the first one did not make

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Monday, January 17, 2005 1:57 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Jamaica


G'Day,
 Can any one recommend a reliable source for DID's for Jamaica?

Thanks
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[Asterisk-Users] Installing * on fedora 3

G'Day List,

Can someone help me out a bit please.
I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying
to install *
I am following
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
After running:
   cd /usr/src/asterisk 
   make clean 
   make 
   make install 
   make samples
The instructions says:
Configuring Asterisk 

- Login to your server as user root 
- Right-click on the background and select Open Terminal 
- Run the following commands to download the VoicePulse Connect! public
key (needed for receiving calls): 

   cd /var/lib/asterisk/keys 
   wget http://connect.voicepulse.com/keys/voicepulse01.pub 

However there is NO /var/lib/asterisk/keys directory.
HELP!!

Thanks



Michael E. Ferguson
Manager, Information Systems
Berman Rennert Vogel  Mandler, P.A.
100 SE 2nd., Street, Suite 2900
Miami, FL., 33131
305.423.3408 Direct
305.533.1582 Fax
[EMAIL PROTECTED] 
This message is for the named person's use only. It may contain
confidential, proprietary or legally privileged information. No
confidentiality or privilege is waived or lost by any mistransmission.
If you receive this message in error, please immediately delete it and
all copies of it from your system, destroy any hard copies of it and
notify the sender. You must not, directly or indirectly, use, disclose,
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right to monitor all e-mail communications through its networks. Any
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RE: [Asterisk-Users] Installing * on fedora 3

I did see an Errors 1.
I will go back and check. Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ishmael
Sent: Tuesday, January 11, 2005 10:34 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Installing * on fedora 3

I run * on FC3 and I have a /var/lib/asterisk/keys directory.  Did the
make of the * software have any errors.

-Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Tuesday, January 11, 2005 9:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Installing * on fedora 3

G'Day List,

Can someone help me out a bit please.
I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying
to install * I am following
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
After running:
   cd /usr/src/asterisk 
   make clean 
   make 
   make install 
   make samples
The instructions says:
Configuring Asterisk 

- Login to your server as user root 
- Right-click on the background and select Open Terminal 
- Run the following commands to download the VoicePulse Connect! public
key (needed for receiving calls): 

   cd /var/lib/asterisk/keys 
   wget http://connect.voicepulse.com/keys/voicepulse01.pub 

However there is NO /var/lib/asterisk/keys directory.
HELP!!

Thanks

Michael E. Ferguson
Manager, Information Systems
Berman Rennert Vogel  Mandler, P.A.
100 SE 2nd., Street, Suite 2900
Miami, FL., 33131
305.423.3408 Direct
305.533.1582 Fax
[EMAIL PROTECTED] 
This message is for the named person's use only. It may contain
confidential, proprietary or legally privileged information. No
confidentiality or privilege is waived or lost by any mistransmission.
If you receive this message in error, please immediately delete it and
all copies of it from your system, destroy any hard copies of it and
notify the sender. You must not, directly or indirectly, use, disclose,
distribute, print, or copy any part of this message if you are not the
intended recipient. BERMAN RENNERT VOGEL  MANDLER, P.A. reserve the
right to monitor all e-mail communications through its networks. Any
views expressed in this message are those of the individual sender,
except where the message states otherwise and the sender is authorized
to state them to be the views of any such entity. 
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RE: [Asterisk-Users] Installing * on fedora 3

G'Day All,

rpm -q kernel-source returns Package kernel-source is not installed
Where can I find it and install it. Asterisk evidently needs it for a
successful install.

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ishmael
Sent: Tuesday, January 11, 2005 10:41 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Installing * on fedora 3


Not sure if this helps, but here's the instructions I followed for
setting up * on FC3:

http://www.automated.it/guidetoasterisk.htm

See if that helps, perhaps there's a step you missed along the way.

-Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Tuesday, January 11, 2005 9:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Installing * on fedora 3

G'Day List,

Can someone help me out a bit please.
I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying
to install * I am following
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
After running:
   cd /usr/src/asterisk 
   make clean 
   make 
   make install 
   make samples
The instructions says:
Configuring Asterisk 

- Login to your server as user root 
- Right-click on the background and select Open Terminal 
- Run the following commands to download the VoicePulse Connect! public
key (needed for receiving calls): 

   cd /var/lib/asterisk/keys 
   wget http://connect.voicepulse.com/keys/voicepulse01.pub 

However there is NO /var/lib/asterisk/keys directory.
HELP!!

Thanks



Michael E. Ferguson
Manager, Information Systems
Berman Rennert Vogel  Mandler, P.A.
100 SE 2nd., Street, Suite 2900
Miami, FL., 33131
305.423.3408 Direct
305.533.1582 Fax
[EMAIL PROTECTED] 
This message is for the named person's use only. It may contain
confidential, proprietary or legally privileged information. No
confidentiality or privilege is waived or lost by any mistransmission.
If you receive this message in error, please immediately delete it and
all copies of it from your system, destroy any hard copies of it and
notify the sender. You must not, directly or indirectly, use, disclose,
distribute, print, or copy any part of this message if you are not the
intended recipient. BERMAN RENNERT VOGEL  MANDLER, P.A. reserve the
right to monitor all e-mail communications through its networks. Any
views expressed in this message are those of the individual sender,
except where the message states otherwise and the sender is authorized
to state them to be the views of any such entity. 
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RE: [Asterisk-Users] Installing * on fedora 3

Thanks very much

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ishmael
Sent: Tuesday, January 11, 2005 3:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Installing * on fedora 3

Usually you select to install the kernel during the installation of FC3,
but I think you can also do:

up2date --get-source kernel

Here's more info:

http://fedoraforum.org/forum/showthread.php?t=29315

Hope that helps,
Dave

-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, January 11, 2005 3:35 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Installing * on fedora 3

G'Day All,

rpm -q kernel-source returns Package kernel-source is not installed
Where can I find it and install it. Asterisk evidently needs it for a
successful install.

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ishmael
Sent: Tuesday, January 11, 2005 10:41 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Installing * on fedora 3

Not sure if this helps, but here's the instructions I followed for
setting up * on FC3:

http://www.automated.it/guidetoasterisk.htm

See if that helps, perhaps there's a step you missed along the way.

-Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Tuesday, January 11, 2005 9:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Installing * on fedora 3

G'Day List,

Can someone help me out a bit please.
I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying
to install * I am following
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
After running:
   cd /usr/src/asterisk 
   make clean 
   make 
   make install 
   make samples
The instructions says:
Configuring Asterisk 

- Login to your server as user root 
- Right-click on the background and select Open Terminal 
- Run the following commands to download the VoicePulse Connect! public
key (needed for receiving calls): 

   cd /var/lib/asterisk/keys 
   wget http://connect.voicepulse.com/keys/voicepulse01.pub 

However there is NO /var/lib/asterisk/keys directory.
HELP!!

Thanks

Michael E. Ferguson
Manager, Information Systems
Berman Rennert Vogel  Mandler, P.A.
100 SE 2nd., Street, Suite 2900
Miami, FL., 33131
305.423.3408 Direct
305.533.1582 Fax
[EMAIL PROTECTED] 
This message is for the named person's use only. It may contain
confidential, proprietary or legally privileged information. No
confidentiality or privilege is waived or lost by any mistransmission.
If you receive this message in error, please immediately delete it and
all copies of it from your system, destroy any hard copies of it and
notify the sender. You must not, directly or indirectly, use, disclose,
distribute, print, or copy any part of this message if you are not the
intended recipient. BERMAN RENNERT VOGEL  MANDLER, P.A. reserve the
right to monitor all e-mail communications through its networks. Any
views expressed in this message are those of the individual sender,
except where the message states otherwise and the sender is authorized
to state them to be the views of any such entity. 
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RE: [Asterisk-Users] Call on hold disconnects...

2004-12-18 Thread Ferguson, Michael

Hang up by taking the call off hold and then hanging up.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wilson
Pickett
Sent: Saturday, December 18, 2004 3:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call on hold disconnects...

I just tried the BT100 hold button the way you described, and that's the
behavior. As someone else noted, how would you ever hang up otherwise?

The behavior you want will be accomplished using parking. However, if no
one retrieves the parked call, it will ring back your phone after a
configurable delay. (see features.conf)
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RE: [Asterisk-Users] Call on hold disconnects...

2004-12-18 Thread Ferguson, Michael

Title: Message

Shoval,
Thanks
much. I will give it a try. 'preciate it.

-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shoval
TomerSent: Saturday, December 18, 2004 9:46 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Call on hold disconnects...

Dialing #700 is, in
fact, parking the call.

That mainly used when
you want to transfer the call to yourself, at another
extension.

Say you pickup the
call In your office, and want to continue it in the server
room.
You can transfer it
there, run over, pick up, then run back and hang up and go back to the server
room and continue the phone call (if you not out of breath
-J)

If you use parking,
you dial #700 and hangup.
Go to the server
room, pickup and dial the extension number you got from the park app and
resume talking.

From:
Christopher Dobbs [mailto:[EMAIL PROTECTED] Sent: Saturday, December 18, 2004 12:03
AMTo: Asterisk Users Mailing
List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Call on
hold disconnects...

Are you having the phone place the person on hold, or
are you having * place them on hold?I dial #700 and it puts them on hold
and they stay there,it also reads off to me the number I dial to get them
off hold.REF: /etc/asterisk/features.conf--Christopher
DobbsShoval Tomer wrote: That's both true and false.We have a legacy PBX here. Panasonic make.Analog extensions connected to it (a.k.a "stupid" extensions) behace exactly like the grandstream - you can put a call on hold, but if you put the handset back on the cradle it's bye bye Mary.Digital extensions (a.k.a "smart" extensions) can hold a call indefinitely.They can do other neat stuff too...
-Original Message-From: Ferguson, Michael [mailto:[EMAIL PROTECTED]]Sent: Friday, December 17, 2004 11:11 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Call on hold disconnects...Antony,Thanks. It seems that the GS will not keep the call on hold.In the real world though, when you place a call on hold, it is held untilfurther action.The caller will hear messages, music, anything while you are gone to lookfor a file, etc.Technically, if you place the call on hold and put the handset back on thecradle, you DID NOT HANG UP to end the call.If you want to hang up the call you will first have to take the call offhold... No.-Original Message-From: [EMAIL PROTECTED] [mailto:asterisk-users-[EMAIL PROTECTED]] On Behalf Of Antony StoneSent: Friday, December 17, 2004 3:54 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Call on hold disconnects...On Friday 17 December 2004 20:43, Ferguson, Michael wrote:
OK. I guess I was not clear. Sorry.The phone rings.The person picks up the handset and speaks to the caller.He then puts the call on hold by pressing the "HOLD" button on the GS100 phone. The caller hears music on hold. So far, so good.
The hand set is placed back on the cradle (as is done on a regularphone with a hold button) I'm not sure I agree with this. Some phones may allow you to hang up andnotdisconnect the call, but I don't think it's universal. Some phonesinterpret this to mean "oh, you want to hang up? Okay - I'll hang up thecallthen."
The call is disconnected. Well, yes, because you hung up.What happens if you do something else, like dial another extension, orpressthe hold button again (perhaps to retreive the original caller)?I repeat one of my original questions - if this is not what you expectedtohappen when you hang up the phone, how would you expect to hang up thecallwhen you wanted to?Antony.--This email is intended for the use of the individual addressee(s) namedaboveand may contain information that is confidential, privileged or unsuitablefor overly sensitive persons with low self-esteem, no sense of humour, orirrational religious beliefs.If you have received this email in error, you are required to shred itimmediately, add some nutmeg, three egg whites and a dessertspoonful ofcaster sugar. Whisk until soft peaks form, then place in a warm oven for40minutes. Remove promptly and let stand for 2 hours before adding somedecorative kiwi fruit and cream. Then notify me immediately by returnemailand eat the original message. Please reply to thelist; please don't CCme. ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listin

RE: [Asterisk-Users] MusicOnHold. not getting it.-GOT IT!!

Mark,
Got it. Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Thursday, December 16, 2004 6:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MusicOnHold. not getting it.

This is well documented in the WIKI.

And, it's not configured from the extensions.conf file. Look for
musiconhold.conf in /etc/asterisk. Genreally one doesn't have to mess
with it but you can do all sorts of neat tricks with it. I have our
office one doing different hold music for different departments so that
they can have their own messages etc played when someone is on hold.

If you want to her your hold music ad a line like this to your
extensions.conf

exten = ,1,Musiconhold(default)

Which will play all the hold music until you hangup.

If you want to play the hold music to peeps while your phone is ringing
do this;

exten = ,1,Dial(SIP/|20|m)

which will play the music for 20 seconds whilst ringing the phone.

Mark

On Thu, 2004-12-16 at 16:57, Ferguson, Michael wrote:

 G'Day All;

 I am a little unsure on how to get Music On Hold to work. Please 
 critique my extensions.conf.  ? Thanks

 ; SIP 5001

 exten = 5001,1,Dial(SIP/5001)

 exten = 5001,2,Voicemail(u${EXTEN})

 exten = 5001,3,Hangup

 exten = 5001,102,Voicemail(b${EXTEN})

 exten = 5001,103,Hangup

 Thanks

 __
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-- 

Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
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[Asterisk-Users] Call on hold disconnects...

G'Day All,

How do I fix this:

I receive a call at the extension. Press the hold button. Music on hold
starts. When I place the handset back on the cradle, the call gets hung
up/disconnected. The Phone is A GrandStream Budge Tone 100.

Thanks


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RE: [Asterisk-Users] Call on hold disconnects...

OK. I guess I was not clear. Sorry.

The phone rings.
The person picks up the handset and speaks to the caller.
He then puts the call on hold by pressing the HOLD button on the GS
100 phone.
The caller hears music on hold.
The hand set is placed back on the cradle (as is done on a regular phone
with a hold button)
The call is disconnected.

Is this normal on a IP phone? I think not.
Does this mean that the GS100 does not really place the call on hold?

Thanks for your feedback.




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antony
Stone
Sent: Friday, December 17, 2004 3:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call on hold disconnects...


On Friday 17 December 2004 20:24, Ferguson, Michael wrote:

 G'Day All,

 How do I fix this:

 I receive a call at the extension. Press the hold button. Music on 
 hold starts. When I place the handset back on the cradle, the call 
 gets hung up/disconnected. The Phone is A GrandStream Budge Tone 100.

1. What would you _expect_ to happen when you do this?
2. If this is a problem, then don't hang up the phone?
3. If you don't want this to happen, how _would_ you hang up if that was
what 
you did want to happen?

Antony.

-- 
I don't know, maybe if we all waited then cosmic rays would write all
our 
software for us. Of course it might take a while.

 - Ron Minnich, Los Alamos National Laboratory

 Please reply to the
list;
   please don't
CC me. ___
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RE: [Asterisk-Users] Call on hold disconnects...

Nabeel,
Thanks.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Friday, December 17, 2004 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call on hold disconnects...

[EMAIL PROTECTED] wrote:
 The caller hears music on hold.
 The hand set is placed back on the cradle (as is done on a regular 
 phone with a hold button) The call is disconnected.

 Is this normal on a IP phone? I think not.
 Does this mean that the GS100 does not really place the call on hold?

http://www.voip-info.org/tiki-print.php?page=Budgetone

This is normal behavior for the phone under all firmware versions; if
you hang up after pressing the Hold button, the call will be
disconnected. (This issue has been brought to Grandstream's attention,
but it is unknown if it will be changed.)

-- 
Nabeel Jafferali
tel: 647.722.8457 x201
 718.606.4190 x201
fwd: 46990 x201
email/msn: nabeelatjafferali.net
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RE: [Asterisk-Users] Call on hold disconnects...

So, the GS is out.
Any recommendations for a Polycom dealer?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Friday, December 17, 2004 3:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call on hold disconnects...


Nabeel,
Thanks.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Friday, December 17, 2004 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call on hold disconnects...


[EMAIL PROTECTED] wrote:
 The caller hears music on hold.
 The hand set is placed back on the cradle (as is done on a regular
 phone with a hold button) The call is disconnected.
 
 Is this normal on a IP phone? I think not.
 Does this mean that the GS100 does not really place the call on hold?

http://www.voip-info.org/tiki-print.php?page=Budgetone

This is normal behavior for the phone under all firmware versions; if
you hang up after pressing the Hold button, the call will be
disconnected. (This issue has been brought to Grandstream's attention,
but it is unknown if it will be changed.)

-- 
Nabeel Jafferali
tel: 647.722.8457 x201
 718.606.4190 x201
fwd: 46990 x201
email/msn: nabeelatjafferali.net
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RE: [Asterisk-Users] Call on hold disconnects...

Antony,
Thanks. It seems that the GS will not keep the call on hold.
In the real world though, when you place a call on hold, it is held until 
further action.
The caller will hear messages, music, anything while you are gone to look for a 
file, etc.

Technically, if you place the call on hold and put the handset back on the 
cradle, you DID NOT HANG UP to end the call.
If you want to hang up the call you will first have to take the call off 
hold... No.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antony Stone
Sent: Friday, December 17, 2004 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call on hold disconnects...


On Friday 17 December 2004 20:43, Ferguson, Michael wrote:

 OK. I guess I was not clear. Sorry.

 The phone rings.
 The person picks up the handset and speaks to the caller.
 He then puts the call on hold by pressing the HOLD button on the GS 
 100 phone. The caller hears music on hold.

So far, so good.

 The hand set is placed back on the cradle (as is done on a regular 
 phone with a hold button)

I'm not sure I agree with this.   Some phones may allow you to hang up and not 
disconnect the call, but I don't think it's universal.   Some phones 
interpret this to mean oh, you want to hang up? Okay - I'll hang up the call 
then.

 The call is disconnected.

Well, yes, because you hung up.

What happens if you do something else, like dial another extension, or press 
the hold button again (perhaps to retreive the original caller)?

I repeat one of my original questions - if this is not what you expected to 
happen when you hang up the phone, how would you expect to hang up the call 
when you wanted to?

Antony.

-- 
This email is intended for the use of the individual addressee(s) named above 
and may contain information that is confidential, privileged or unsuitable 
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RE: [Asterisk-Users] Call on hold disconnects...

thanks

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer
Sent: Friday, December 17, 2004 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call on hold disconnects...

Also, you can always park the call instead of holding it.

 -Original Message-
 From: Ferguson, Michael [mailto:[EMAIL PROTECTED]
 Sent: Friday, December 17, 2004 11:11 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Call on hold disconnects...

 Antony,
 Thanks. It seems that the GS will not keep the call on hold. In the 
 real world though, when you place a call on hold, it is held until 
 further action. The caller will hear messages, music, anything while 
 you are gone to look for a file, etc.

 Technically, if you place the call on hold and put the handset back on 
 the cradle, you DID NOT HANG UP to end the call. If you want to hang 
 up the call you will first have to take the call off hold... No.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users- 
 [EMAIL PROTECTED] On Behalf Of Antony Stone
 Sent: Friday, December 17, 2004 3:54 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Call on hold disconnects...

 On Friday 17 December 2004 20:43, Ferguson, Michael wrote:

  OK. I guess I was not clear. Sorry.

  The phone rings.
  The person picks up the handset and speaks to the caller. He then 
  puts the call on hold by pressing the HOLD button on the GS 100 
  phone. The caller hears music on hold.

 So far, so good.

  The hand set is placed back on the cradle (as is done on a regular 
  phone with a hold button)

 I'm not sure I agree with this.   Some phones may allow you to hang up and
 not
 disconnect the call, but I don't think it's universal.   Some phones
 interpret this to mean oh, you want to hang up? Okay - I'll hang up 
 the call then.

  The call is disconnected.

 Well, yes, because you hung up.

 What happens if you do something else, like dial another extension, or 
 press the hold button again (perhaps to retreive the original caller)?

 I repeat one of my original questions - if this is not what you 
 expected to happen when you hang up the phone, how would you expect to 
 hang up the call
 when you wanted to?

 Antony.

 --
 This email is intended for the use of the individual addressee(s) 
 named above and may contain information that is confidential, 
 privileged or unsuitable for overly sensitive persons with low 
 self-esteem, no sense of humour, or irrational religious beliefs.

 If you have received this email in error, you are required to shred it 
 immediately, add some nutmeg, three egg whites and a dessertspoonful 
 of caster sugar.   Whisk until soft peaks form, then place in a warm 
 oven for 40 minutes.   Remove promptly and let stand for 2 hours 
 before adding some decorative kiwi fruit and cream.   Then notify me 
 immediately by return email
 and eat the original message.

  Please reply to 
 the list;
please 
 don't CC me. ___
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RE: [Asterisk-Users] Call on hold disconnects...

Title: Message



I am 
pressing the HOLD button on the GS phone

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  Christopher DobbsSent: Friday, December 17, 2004 5:03 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [Asterisk-Users] Call on hold 
  disconnects...Are you having the phone place the person 
  on hold, or are you having * place them on hold?I dial #700 and it puts 
  them on hold and they stay there,it also reads off to me the number I dial 
  to get them off hold.REF: 
  /etc/asterisk/features.conf--Christopher DobbsShoval Tomer 
  wrote: 
  That's both true and false.

We have a legacy PBX here. Panasonic make.
Analog extensions connected to it (a.k.a "stupid" extensions) behace exactly like the grandstream - you can put a call on hold, but if you put the handset back on the cradle it's bye bye Mary.

Digital extensions (a.k.a "smart" extensions) can hold a call indefinitely.
They can do other neat stuff too...

 

  
-Original Message-----
From: Ferguson, Michael [mailto:[EMAIL PROTECTED]]
Sent: Friday, December 17, 2004 11:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call on hold disconnects...

Antony,
Thanks. It seems that the GS will not keep the call on hold.
In the real world though, when you place a call on hold, it is held until
further action.
The caller will hear messages, music, anything while you are gone to look
for a file, etc.

Technically, if you place the call on hold and put the handset back on the
cradle, you DID NOT HANG UP to end the call.
If you want to hang up the call you will first have to take the call off
hold... No.

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]] On Behalf Of Antony Stone
Sent: Friday, December 17, 2004 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call on hold disconnects...


On Friday 17 December 2004 20:43, Ferguson, Michael wrote:


  OK. I guess I was not clear. Sorry.

The phone rings.
The person picks up the handset and speaks to the caller.
He then puts the call on hold by pressing the "HOLD" button on the GS
100 phone. The caller hears music on hold.
  So far, so good.


  The hand set is placed back on the cradle (as is done on a regular
phone with a hold button)
  I'm not sure I agree with this.   Some phones may allow you to hang up and
not
disconnect the call, but I don't think it's universal.   Some phones
interpret this to mean "oh, you want to hang up? Okay - I'll hang up the
call
then."


  The call is disconnected.
  Well, yes, because you hung up.

What happens if you do something else, like dial another extension, or
press
the hold button again (perhaps to retreive the original caller)?

I repeat one of my original questions - if this is not what you expected
to
happen when you hang up the phone, how would you expect to hang up the
call
when you wanted to?

Antony.

--
This email is intended for the use of the individual addressee(s) named
above
and may contain information that is confidential, privileged or unsuitable
for overly sensitive persons with low self-esteem, no sense of humour, or
irrational religious beliefs.

If you have received this email in error, you are required to shred it
immediately, add some nutmeg, three egg whites and a dessertspoonful of
caster sugar.  Whisk until soft peaks form, then place in a warm oven for
40
minutes.  Remove promptly and let stand for 2 hours before adding some
decorative kiwi fruit and cream.  Then notify me immediately by return
email
and eat the original message.

 Please reply to the
list;
   please don't CC
me. ___
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[Asterisk-Users] MusicOnHold. not getting it.

2004-12-16 Thread Ferguson, Michael

Title: Message





G'Day 
All;

I am a 
little unsure on how to get Music On Hold to work. Please critique my 
extensions.conf. ? Thanks


; SIP 5001
exten = 5001,1,Dial(SIP/5001)
exten = 5001,2,Voicemail(u${EXTEN})
exten = 5001,3,Hangup
exten = 5001,102,Voicemail(b${EXTEN})
exten = 5001,103,Hangup


Thanks
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RE: [Asterisk-Users] MusicOnHold. not getting it.

2004-12-16 Thread Ferguson, Michael

Mark, Thanks for the pointer. 'preciate it.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Thursday, December 16, 2004 6:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MusicOnHold. not getting it.

This is well documented in the WIKI.

And, it's not configured from the extensions.conf file. Look for
musiconhold.conf in /etc/asterisk. Genreally one doesn't have to mess
with it but you can do all sorts of neat tricks with it. I have our
office one doing different hold music for different departments so that
they can have their own messages etc played when someone is on hold.

If you want to her your hold music ad a line like this to your
extensions.conf

exten = ,1,Musiconhold(default)

Which will play all the hold music until you hangup.

If you want to play the hold music to peeps while your phone is ringing
do this;

exten = ,1,Dial(SIP/|20|m)

which will play the music for 20 seconds whilst ringing the phone.

Mark

On Thu, 2004-12-16 at 16:57, Ferguson, Michael wrote:

 G'Day All;

 I am a little unsure on how to get Music On Hold to work. Please 
 critique my extensions.conf.  ? Thanks

 ; SIP 5001

 exten = 5001,1,Dial(SIP/5001)

 exten = 5001,2,Voicemail(u${EXTEN})

 exten = 5001,3,Hangup

 exten = 5001,102,Voicemail(b${EXTEN})

 exten = 5001,103,Hangup

 Thanks

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-- 

Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
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RE: [Asterisk-Users] Ethernet Channel Bank (Comming Soon to a NOC NearYou!)

2004-12-13 Thread Ferguson, Michael

Curious here,

What does that mean. Thanks



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Christopher Dobbs
Sent: Monday, December 13, 2004 8:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Ethernet Channel Bank (Comming Soon to a NOC
NearYou!)


My company has started development on a Ethernet based channel bank.

Here are the (current) spec's
- 10/100 Ethernet Port
- Up to 96 FXS/FXO ports (Thats 4 DS1's for the math impaired)
- Serial Console
- TDMoE
- IAX2
- EETP (A protocol that we have designed for IP Telephony)

We have just started prototyping this device, so...

--
Christopher Dobbs

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RE: [Asterisk-Users] Asterisk Training Needed in SouthEast U.S

2004-12-10 Thread Ferguson, Michael

Title: Message



Me 
too. For the South Fla area.

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Paul 
  RodanSent: Friday, December 10, 2004 4:31 PMTo: 
  'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
  [Asterisk-Users] Asterisk Training Needed in SouthEast 
U.S
  
  I need advanced Asterisk training 
  in the SouthEast area of the U.S; I dont need to know how to install linux 
  and Asterisk and compile the modules and load them and such. I dont need to 
  know what extensions.conf does or sip.conf does; 
  
  What I do need is a better 
  understanding of what every single little option in sip.conf or iax.conf does, 
  and I need to learn a lot more about all the tips and tricks and neat little 
  things you can do in the extensions.conf file. I also want to increase my 
  knowledge of SIP and IAX in general, and maybe learn a little about Zaptel 
  cards and how to use them. Seems everytime I go over the sip.conf file, a 
  month or two later when I get the latest CVS, new options and change of 
  default parameters appear in the example configs. 

  
  Anybody know of a good training 
  center or tutor in this area that can help train me? I intend to make the 
  company I work for pay for it, as they seem content to stick me with being the 
  only VOIP Admin. Ive learned a lot since I started 3 months ago, but every 
  time I start browsing the Wiki I feel like my head is going to explode, so 
  much information to absorb and so many confusing terms. Im more of a hands on 
  and learn-by-examples person. Most of what Ive learned is by reverse 
  engineering the config files left behind by the previous VOIP Admin who was 
  real good but no longer working here and no longer willing to assist. I found 
  an interesting 3-day course offered by this one company, but they turned out 
  to be in London, and itll be tough enough to get them 
  to transport and pay for this course in a local area, trying to get them to 
  fly me out of the U.S is infeasible. 

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RE: [Asterisk-Users] Ripping CD audio for MOH

2004-12-10 Thread Ferguson, Michael

Brian,
Can you then please expound on the best way to go about getting MOH
setup with the files. Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Friday, December 10, 2004 4:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Ripping CD audio for MOH

No I know exactly how to use lame. :P  In my opinion 32k doesn't sound
the same as 128k tracks that get down sampled to be played. 

I'm not known for being eloquent... because I don't beat around the
bush. :P (some might get this joke)

This is why I don't really think MP3's are the answer:
http://bugs.digium.com/bug_view_page.php?bug_id=0002639

bkw
PS: Our company is the one that wrote that patch.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users- 
 [EMAIL PROTECTED] On Behalf Of Jay Milk
 Sent: Friday, December 10, 2004 3:32 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Ripping CD audio for MOH

 Thanks for the very eloquent reply.  My MOH music is of the classical 
 genre and sounds excellent.  It sounds so good, that I've already had 
 another business ask me if I could set up their MOH.  Maybe I got 
 lucky... Or maybe you don't know how to use lame?

  -Original Message-
  From: Brian West [mailto:[EMAIL PROTECTED]
  Sent: Friday, December 10, 2004 3:07 PM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: RE: [Asterisk-Users] Ripping CD audio for MOH

  That's the worst thing you could EVER do because you'll introduce 
  compression artifacts into the hold music and it will sound like 
  TOTAL ASS.

  bkw

   -Original Message-
   From: [EMAIL PROTECTED]
 [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Jay Milk
  Sent: Friday, December 10, 2004 2:50 PM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: RE: [Asterisk-Users] Ripping CD audio for MOH

  I used existing mp3s and recoded them using lame to mono, 32kbps or 
  thereabouts.

   -Original Message-
   From: Thomas Johnson [mailto:[EMAIL PROTECTED]
   Sent: Friday, December 10, 2004 12:01 PM
   To: [EMAIL PROTECTED]
   Subject: [Asterisk-Users] Ripping CD audio for MOH

   Hello-

   I've got some audio CDs that I'd like to use for MOH.

   What's the best way to do this?  I don't care if it's mp3 or some 
   other format - whatever will work best.

   What applications (osx or linux) are best?  Optimal settings?

   Thanks-

   Tom

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RE: [Asterisk-Users] Ripping CD audio for MOH

2004-12-10 Thread Ferguson, Michael

Thanks Brian

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Friday, December 10, 2004 5:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Ripping CD audio for MOH


You can use sox to convert them to ulaw (which is what I would
recommend)

bkw

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users- 
 [EMAIL PROTECTED] On Behalf Of Ferguson, Michael
 Sent: Friday, December 10, 2004 3:58 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Ripping CD audio for MOH
 
 Brian,
 Can you then please expound on the best way to go about getting MOH 
 setup with the files. Thanks
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Brian 
 West
 Sent: Friday, December 10, 2004 4:46 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Ripping CD audio for MOH
 
 
 No I know exactly how to use lame. :P  In my opinion 32k doesn't sound

 the same as 128k tracks that get down sampled to be played.
 
 I'm not known for being eloquent... because I don't beat around the 
 bush. :P (some might get this joke)
 
 This is why I don't really think MP3's are the answer: 
 http://bugs.digium.com/bug_view_page.php?bug_id=0002639
 
 bkw
 PS: Our company is the one that wrote that patch.
 
 
  -Original Message-
  From: [EMAIL PROTECTED] 
  [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jay 
  Milk
  Sent: Friday, December 10, 2004 3:32 PM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: RE: [Asterisk-Users] Ripping CD audio for MOH
 
  Thanks for the very eloquent reply.  My MOH music is of the 
  classical genre and sounds excellent.  It sounds so good, that I've 
  already had another business ask me if I could set up their MOH.  
  Maybe I got lucky... Or maybe you don't know how to use lame?
 
   -Original Message-
   From: Brian West [mailto:[EMAIL PROTECTED]
   Sent: Friday, December 10, 2004 3:07 PM
   To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   Subject: RE: [Asterisk-Users] Ripping CD audio for MOH
  
  
   That's the worst thing you could EVER do because you'll introduce 
   compression artifacts into the hold music and it will sound like 
   TOTAL ASS.
  
   bkw
  
-Original Message-
From: [EMAIL PROTECTED]
  [mailto:asterisk-users-
   [EMAIL PROTECTED] On Behalf Of Jay Milk
   Sent: Friday, December 10, 2004 2:50 PM
   To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   Subject: RE: [Asterisk-Users] Ripping CD audio for MOH
  
   I used existing mp3s and recoded them using lame to mono, 32kbps 
   or thereabouts.
  
-Original Message-
From: Thomas Johnson [mailto:[EMAIL PROTECTED]
Sent: Friday, December 10, 2004 12:01 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Ripping CD audio for MOH
   
   
Hello-
   
I've got some audio CDs that I'd like to use for MOH.
   
What's the best way to do this?  I don't care if it's mp3 or 
some other format - whatever will work best.
   
What applications (osx or linux) are best?  Optimal settings?
   
Thanks-
   
Tom
  
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[Asterisk-Users] Voicemail messages by email

G'Day All
How do I configure the mailer on the asterisk box to send a FQDN as a
part of the message so that ISP's doing reverse DNS does not drop the
asterisk mail?

Thanks 


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RE: [Asterisk-Users] Voicemail messages by email

I did that already but it did not work. Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Thursday, December 09, 2004 3:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Voicemail messages by email


serveremail= in voicemail.conf

bkw

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users- 
 [EMAIL PROTECTED] On Behalf Of Ferguson, Michael
 Sent: Thursday, December 09, 2004 1:58 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Voicemail messages by email
 
 G'Day All
 How do I configure the mailer on the asterisk box to send a FQDN as a 
 part of the message so that ISP's doing reverse DNS does not drop the 
 asterisk mail?
 
 Thanks
 
 
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RE: [Asterisk-Users] Voicemail messages by email

Thanks for the reply.
This * box currently sits on a WINDOWS network behind a firewall and
does not have a FQDN.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Thursday, December 09, 2004 3:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Voicemail messages by email

I did that already but it did not work. Thanks

The issue might be the FQDN of the * server itself. If you are using a
smarthost under your control for relay, the host name should be
substituted automatically with the smarthost's FQDN. If not, then in
your
/etc/mail/sendmail.cf:

DSmachine name with FQDN
Dj$wdomain-name

example:

DS foo.somewhere.com
Dj$w.somewhere.com

I think there is an example of this commented out in sendmail.cf

hth
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RE: [Asterisk-Users] Voicemail messages by email

Yeah. I am working on trying it. Thanks again. I will let you know

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Thursday, December 09, 2004 4:08 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Voicemail messages by email

Thanks for the reply.
This * box currently sits on a WINDOWS network behind a firewall and 
does not have a FQDN.

Still should work. Try it. Sendmail attempts to resolve it's hostname
against dns and it's own hosts entries, and if it can't, it puts in
whatever it can, even localhost. The DS foo.somewhere.com and
Dj$w.somewhere.com overrides that behavior if resolution fails. 
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RE: [Asterisk-Users] --SOLVED--Voicemail messages by email

Colin,
Thanks very much for your feedback. I have achieved success.
I appreciate it. Thank you and thnaks to the List.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Thursday, December 09, 2004 4:08 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Voicemail messages by email

Thanks for the reply.
This * box currently sits on a WINDOWS network behind a firewall and 
does not have a FQDN.

Still should work. Try it. Sendmail attempts to resolve it's hostname
against dns and it's own hosts entries, and if it can't, it puts in
whatever it can, even localhost. The DS foo.somewhere.com and
Dj$w.somewhere.com overrides that behavior if resolution fails. 
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RE: [Asterisk-Users] Why, why, why???

2004-12-06 Thread Ferguson, Michael

Thanks very much. I will give it a try.
'preciate it.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Sent: Sunday, December 05, 2004 4:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Why, why, why???


On Fri, 2004-12-03 at 16:54 -0500, Ferguson, Michael wrote:
 [incoming]
 exten = 321XXX,1,Goto(incoming,s,1)

Afaik all regex numbers should start with an underscore so that should
read _321XXX I guess.

[snip]
 
 SIP.CONF
 [general]
 port=5060
 bindaddr=0.0.0.0  ; IP address to bind to (0.0.0.0 binds
 to all)
 externip=XXX.XXX.XXX.XXX
 localnet=192.168.131.0
 localmask=255.255.255.0
 context=incoming
 tos=lowdelay
 disallow=all
 allow=ulaw
 context=invalid

You have a context in here twice. That looks like one too many.

Regards,
Patrick

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RE: [Asterisk-Users] Why, why, why???

2004-12-04 Thread Ferguson, Michael

Noah,
Thanks for the reply. I will try your instructions on Monday. I
appreciate it very much

Ferg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Miller
Sent: Friday, December 03, 2004 6:16 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Why, why, why???


Hi Michael -

 Thanks very much. See below. I do not have a zaptel.conf

I made the assumption you were using Digium hardware, sorry.  What 
device are you using for your incoming lines?

For the fast busy:

 [incoming]
 exten = 321XXX,1,Goto(incoming,s,1)
 exten = s,1,Answer
 exten = s,2,DigitTimeout(10)
 exten = s,3,ResponseTimeout(20)
 exten = s,4,Background(swelcome)
 exten = t,1,Hangup
 include =extensions

Are you dialing in on one of the 321XXX lines, or another number?


For the one way audio on the grandstream:

 [5001]
 type=friend   ; either friend (peer+user), peer or
user
 host=dynamic
 username=5001
 context=toll-access
 canreinvite=no
 quality=300
 callerid=5001
 disallow=all
 allow=ulaw
 allow=alaw
 [EMAIL PROTECTED]
 nat=no
 dtmfmode=rfc2833

It looks like it should work, but I don't use grandstream phones.  Has 
anybody else had this problem?  Have you tried the latest version of 
the Grandstream firmware - I know older versions had a number of 
problems.

Thanks,
Noah

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RE: [Asterisk-Users] Why, why, why???

2004-12-04 Thread Ferguson, Michael

The * server is behind a Watchguard Firewall and I do have ports
forwarded. I will chyeck them on Monday. Thanks to all.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wilson
Pickett
Sent: Saturday, December 04, 2004 10:54 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Why, why, why???

 We have Grandstream SIP phones with the latest firmware versions and 
 have also have this problem.  It appears to be something to do with 
 RTP, I believe.  I don't know exactly what (simply because I don't 
 know much about RTP as yet), but the packets don't seem to reach the 
 Grandstream from the other phone.  The phones appear to work correctly

 when located on the same LAN segment.  But, when one is placed behind 
 a NAT router, the dynamic changes and one-way audio seems to happen 
 frequently.  I've

Are you forwarding ports? What ports have you set asterisk to? 

IIRC the GS phones come with 8000 by default and asterisk comes with
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RE: [Asterisk-Users] Why, why, why???

2004-12-04 Thread Ferguson, Michael

I do not have the Digium card on this box.
I have it on another box that I will eventually from it from.
All incoming calls are through IP and not any POTS line

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Miller
Sent: Friday, December 03, 2004 6:16 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Why, why, why???

Hi Michael -

 Thanks very much. See below. I do not have a zaptel.conf

I made the assumption you were using Digium hardware, sorry.  What 
device are you using for your incoming lines?

For the fast busy:

 [incoming]
 exten = 321XXX,1,Goto(incoming,s,1)
 exten = s,1,Answer
 exten = s,2,DigitTimeout(10)
 exten = s,3,ResponseTimeout(20)
 exten = s,4,Background(swelcome)
 exten = t,1,Hangup
 include =extensions

Are you dialing in on one of the 321XXX lines, or another number?

For the one way audio on the grandstream:

 [5001]
 type=friend   ; either friend (peer+user), peer or
user
 host=dynamic
 username=5001
 context=toll-access
 canreinvite=no
 quality=300
 callerid=5001
 disallow=all
 allow=ulaw
 allow=alaw
 [EMAIL PROTECTED]
 nat=no
 dtmfmode=rfc2833

It looks like it should work, but I don't use grandstream phones.  Has 
anybody else had this problem?  Have you tried the latest version of 
the Grandstream firmware - I know older versions had a number of 
problems.

Thanks,
Noah

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[Asterisk-Users] Fast Busy

2004-12-03 Thread Ferguson, Michael

Title: Message



G'Day 
All,
Can I get a little 
help here? Thanks.
I just completed an 
* setup. I have a GS Budgetone 101.
I can call out 
ok.
When I call the 
phone number to the * server I get a fast busy.

Any ideas? Thanks 
much.



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[Asterisk-Users] Why, why, why???

2004-12-03 Thread Ferguson, Michael

Help.

Why is it that I can call out from my GSBudgetone SIP phone but the
audio is one-way'?

Why is it that when I call my asterisk phone number, I get a fast busy?


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RE: [Asterisk-Users] Why, why, why???

2004-12-03 Thread Ferguson, Michael

Thanks very much. See below. I do not have a zaptel.conf

**
Extensions.conf
[globals]
[extensions]
;directory app
exten = 9,1,Directory(extensions)

; echo latency test
exten = 10,1,Playback(demo-echotest)
exten = 10,2,Playback(beep)
exten = 10,3,Echo
exten = 10,4,Playback(demo-echodone)
exten = 10,5,Hangup

; 1000Hz tone test
exten = 11,1,Milliwatt()
exten = 11,2,Hangup

; exten for recording greetings/menus
exten = 12,1,Authenticate(12|)
exten = 12,1,Wait(2)
exten = 12,2,Record(/var/lib/asterisk/sounds/swelcome:gsm)
exten = 12,3,Wait(2)
exten = 12,4,Playback(/var/lib/asterisk/sounds/swelcome)
exten = 12,5,Wait(2)
  
exten = 12,6,Hangup

; date and time check
exten = 13,1,DateTime()
exten = 13,2,Wait(1)
exten = 13,3,DateTime()
exten = 13,4,Hangup

; extension check
exten = 14,1,Wait(1)
exten = 14,2,SayDigits(${CALLERIDNUM})
exten = 14,3,Wait(1)
exten = 14,4,SayDigits(${CALLERIDNUM})
exten = 14,5,Hangup

; user's voicemail
exten = 15,1,VoicemailMain
exten = 15,2,Hangup

; SIP 5000
exten = 5000,1,Dial(SIP/5000)
exten = 5000,2,Voicemail(u${EXTEN})
exten = 5000,3,Hangup
exten = 5000,102,Voicemail(b${EXTEN})
exten = 5000,103,Hangup

; SIP 5001
exten = 5001,1,Dial(SIP/5001)
exten = 5001,2,Voicemail(u${EXTEN})
exten = 5001,3,Hangup
exten = 5001,102,Voicemail(b${EXTEN})
exten = 5001,103,Hangup

; MeetMe
exten = 200,1,Answer
exten = 200,2,Wait(1)
;exten = 200,3,Authenticate(109)
exten = 200,3,MeetMe(1|Masp)
exten = 200,4,Playback(vm-goodbye)
exten = 200,5,Hangup

[incoming]
exten = 321XXX,1,Goto(incoming,s,1)
exten = s,1,Answer
exten = s,2,DigitTimeout(10)
exten = s,3,ResponseTimeout(20)
exten = s,4,Background(swelcome)
exten = t,1,Hangup
include =extensions

[toll-trunks];voicepulse for now
 
[voicepulse]
;voice over IP outgoing
exten =
_NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/1${EXTEN
})
exten =
_NXXNXX,102,Dial(IAX2/[EMAIL PROTECTED]/1${EXT
EN})
exten =
_1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN
})
exten =
_1NXXNXX,102,Dial(IAX2/[EMAIL PROTECTED]/${EXT
EN})
exten =
_011.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten =
_011.,102,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})

;911
;exten = 911,1,ChanIsAvail(Zap/1)
exten = 911,1,Dial(Zap/g1/911)
exten = 911,2,Hangup()
exten = 911,102,SoftHangup (Zap/1-1)
exten = 911,103,Wait(1)
exten = 911,104,Goto(1)

;411
exten = 411,1,Dial(Zap/g1/411)
exten = 411,2,Hangup


[local-trunks]

[local-access]
ignorepat = 9
include =extensions
include = local-trunks
include = voicepulse

[toll-access]
ignorepat = 9
include = local-access
include = toll-trunks
include = voicepulse

**

SIP.CONF
[general]
port=5060
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds
to all)
externip=XXX.XXX.XXX.XXX
localnet=192.168.131.0
localmask=255.255.255.0
context=incoming
tos=lowdelay
disallow=all
allow=ulaw
context=invalid

[5001]
type=friend ; either friend (peer+user), peer or
user
host=dynamic
username=5001
context=toll-access
canreinvite=no
quality=300
callerid=5001
disallow=all
allow=ulaw
allow=alaw
[EMAIL PROTECTED]
nat=no
dtmfmode=rfc2833

ZAPATA.CONF


[channels]
;
; Default language
;
;language=en
;
; Default context
;
context=default
;
; Switchtype:  Only used for PRI.
;
; national:   National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess:   ATT 4ESS
; 5ess:   Lucent 5ESS
; euroisdn:   EuroISDN
; ni1:Old National ISDN 1
;
switchtype=national
;
; Some switches (ATT especially) require network specific facility IE
; supported values are currently 'none', 'sdn', 'megacom', 'accunet'
;
;nsf=none
;
; PRI Dialplan:  Only RARELY used for PRI.
;
; unknown:Unknown
; private:Private ISDN
; local:  Local ISDN
; national:   National ISDN
; international:  International ISDN
;
;pridialplan=national
;
; PRI Local Dialplan:  Only RARELY used for PRI (sets the calling
number's numbering plan)
;
; unknown:Unknown
; private:Private ISDN
; local:  Local ISDN
; national:   National ISDN
; international:  International ISDN
;
;prilocaldialplan=national
;
; Overlap dialing mode (sending overlap digits)
;
;overlapdial=yes
;
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to
work
; with all telcos.
; 
; outofband:  Signal Busy/Congestion out of band with
RELEASE/DISCONNECT
; inband: Signal Busy/Congestion using in-band tones
;
; priindication = outofband
;
; ISDN Timers
; All of the ISDN timers and counters that are used are configurable.
Specify 
; the timer name, and its value (in ms for timers)
;
; pritimer = t200,1000
; pritimer = t313,4000
;
;
; Signalling method (default is fxs).  Valid

[Asterisk-Users] Restarting *

2004-12-02 Thread Ferguson, Michael

Title: Message



G'Day 
All

What do I type at 
the command line to stop and start * on a RedHat ES3 box?

Thanks




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RE: [Asterisk-Users] Restarting *

2004-12-02 Thread Ferguson, Michael

Thanks to everyone for the help.

My next problem is that I have no audio.
I have two extensions, 5001 and 5002.
If I dial 5002 from 5001 it rings fine but when it is picked up there is
no autio.
Also if I call into the phone number from outside, I get the answer,
select the extention number, 5001, but it rings endlessly and does not
transfer to Voicemail.

Can you please point me in the right direction. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Dent
Sent: Thursday, December 02, 2004 10:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Restarting *


Probably 

/etc/rc.d/init/asterisk restart

Mike



On Thu, 2 Dec 2004 09:50:51 -0500, Ferguson, Michael
[EMAIL PROTECTED] wrote:
  
 G'Day All
   
 What do I type at the command line to stop and start * on a RedHat ES3

 box?
   
 Thanks
   
   
  
   
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[Asterisk-Users] Almost there--Remote connection

G'Day All;

Greetings and best wishes. I need some help as follows:

My Grandstream 100 is at a remote location on broadband and connects to
my * server else where.
From a POST line I dial the 3 to the * server and selects the ext # of
the remote GS100 IP phone.
The GS100 rings. When answered I can clearly hear everything coming from
the phone that's calling in.
The caller cannot hear anything coming from the GS100 IP phone.

If I make a call out from the GS100 to a POTS #, the POTS number rings.
Upon answering, the GS100 can also hear everything from the POTS phone
but the POTS phone is not hearing anything from the GS100.

I believe the phone is setup right.

The * server is behind a firewall and I have opened ports 
1-10100
5060
5004
4569

So it seems that my something is not allowing signal from the GS100 IP
phone out but is allowing signal in.

Any thoughts one where/what I should be modifying?

Thanks much.  
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RE: [Asterisk-Users] Almost there--Remote connection

The 1-10100 was given to me by a prior post so I really do not know.
I will change the forewall to allow 1-2 and see if it works.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl Dyson
Sent: Tuesday, October 19, 2004 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connection


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ferguson, Michael
 Sent: 19 October 2004 16:18
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Almost there--Remote connection
 

[snip]

 
 The * server is behind a firewall and I have opened ports 
 1-10100 5060
 5004
 4569
 

IIRC, SIP uses 1-2 by default. Have you changed this to
1-10100?

Cheers,

Karl


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RE: [Asterisk-Users] Almost there--Remote connection

Thanks. I think that's Iptables. No?
I have a hardware firewall.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Deon
Rodden
Sent: Tuesday, October 19, 2004 11:35 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Almost there--Remote connection


My firewall script has something to the effect of:

# Allow Existing traffic through
-A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT

# Incoming VOIP Ports
-A INPUT -m state --state NEW -m tcp -p tcp --dport 5036:5045 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 5036:5045 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 2727:2727 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 4569:4569 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 5060:5061 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 1:2 -j
ACCEPT

That's for IAX2 and SIP.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl Dyson
Sent: Tuesday, October 19, 2004 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connection

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ferguson, Michael
 Sent: 19 October 2004 16:18
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Almost there--Remote connection
 

[snip]

 
 The * server is behind a firewall and I have opened ports 
 1-10100 5060
 5004
 4569
 

IIRC, SIP uses 1-2 by default. Have you changed this to
1-10100?

Cheers,

Karl


This e-mail has been scanned for all viruses by Star. The
service is powered by MessageLabs. For more information on a proactive
anti-virus service working around the clock, around the globe, visit:
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RE: [Asterisk-Users] Almost there--Remote connection

I made the firewall changes but still the same result.

On the GS100 phone, what us STUN server?
Why is it important?
If it say No in the config, I hear nothing.
If it says and has GS's STUN IP the connection is one way as noted
prior. Might this be the culprit?

Thanks... I am almost there!! 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Deon
Rodden
Sent: Tuesday, October 19, 2004 11:35 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Almost there--Remote connection


My firewall script has something to the effect of:

# Allow Existing traffic through
-A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT

# Incoming VOIP Ports
-A INPUT -m state --state NEW -m tcp -p tcp --dport 5036:5045 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 5036:5045 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 2727:2727 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 4569:4569 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 5060:5061 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 1:2 -j
ACCEPT

That's for IAX2 and SIP.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl Dyson
Sent: Tuesday, October 19, 2004 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connection

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ferguson, Michael
 Sent: 19 October 2004 16:18
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Almost there--Remote connection
 

[snip]

 
 The * server is behind a firewall and I have opened ports 
 1-10100 5060
 5004
 4569
 

IIRC, SIP uses 1-2 by default. Have you changed this to
1-10100?

Cheers,

Karl


This e-mail has been scanned for all viruses by Star. The
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RE: [Asterisk-Users] Almost there--Remote connection

Thanks. 

Mine says
rtpstart=1
rtpend=2






-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl Dyson
Sent: Tuesday, October 19, 2004 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connection


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ferguson, Michael
 Sent: 19 October 2004 16:49
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Almost there--Remote connection
 
 Thanks. I think that's Iptables. No?
 I have a hardware firewall.

First, have a peek in rtp.conf and see what it says its using. For
example, my (modified) version looks like:

;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=15000
rtpend=17000

 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Deon Rodden
 Sent: Tuesday, October 19, 2004 11:35 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Almost there--Remote connection
 
 
 My firewall script has something to the effect of:
 
 # Allow Existing traffic through
 -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT
 
 # Incoming VOIP Ports
 -A INPUT -m state --state NEW -m tcp -p tcp --dport 5036:5045
 -j ACCEPT -A INPUT -m state --state NEW -m udp -p udp --dport 
 5036:5045 -j ACCEPT -A INPUT -m state --state NEW -m udp -p 
 udp --dport 2727:2727 -j ACCEPT -A INPUT -m state --state NEW 
 -m udp -p udp --dport 4569:4569 -j ACCEPT -A INPUT -m state 
 --state NEW -m udp -p udp --dport 5060:5061 -j ACCEPT -A 
 INPUT -m state --state NEW -m udp -p udp --dport 1:2 -j ACCEPT
 
 That's for IAX2 and SIP.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Karl Dyson
 Sent: Tuesday, October 19, 2004 11:22 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Almost there--Remote connection
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  Ferguson, Michael
  Sent: 19 October 2004 16:18
  To: [EMAIL PROTECTED]
  Subject: [Asterisk-Users] Almost there--Remote connection
  
 
 [snip]
 
  
  The * server is behind a firewall and I have opened ports
 1-10100
  5060
  5004
  4569
  
 
 IIRC, SIP uses 1-2 by default. Have you changed this
 to 1-10100?
 
 Cheers,
 
 Karl
 
 __
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 proactive anti-virus service working around the clock, around 
 the globe, visit:
 http://www.star.net.uk
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RE: [Asterisk-Users] Almost there--Remote connection

I just realised that I neglected to mention that the remote GS100 phone
is sitting behind a firewall also.
Do I need to open any outgoing ports on that firewall? Considering that
one cannot hear anything from the GS100 IP phone?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Tuesday, October 19, 2004 12:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connection


Thanks. 

Mine says
rtpstart=1
rtpend=2






-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl Dyson
Sent: Tuesday, October 19, 2004 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connection


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Ferguson, Michael
 Sent: 19 October 2004 16:49
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Almost there--Remote connection
 
 Thanks. I think that's Iptables. No?
 I have a hardware firewall.

First, have a peek in rtp.conf and see what it says its using. For
example, my (modified) version looks like:

;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=15000
rtpend=17000

 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Deon 
 Rodden
 Sent: Tuesday, October 19, 2004 11:35 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Almost there--Remote connection
 
 
 My firewall script has something to the effect of:
 
 # Allow Existing traffic through
 -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT
 
 # Incoming VOIP Ports
 -A INPUT -m state --state NEW -m tcp -p tcp --dport 5036:5045 -j 
 ACCEPT -A INPUT -m state --state NEW -m udp -p udp --dport 5036:5045 
 -j ACCEPT -A INPUT -m state --state NEW -m udp -p udp --dport 
 2727:2727 -j ACCEPT -A INPUT -m state --state NEW -m udp -p udp 
 --dport 4569:4569 -j ACCEPT -A INPUT -m state --state NEW -m udp -p 
 udp --dport 5060:5061 -j ACCEPT -A INPUT -m state --state NEW -m udp 
 -p udp --dport 1:2 -j ACCEPT
 
 That's for IAX2 and SIP.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Karl 
 Dyson
 Sent: Tuesday, October 19, 2004 11:22 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Almost there--Remote connection
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  Ferguson, Michael
  Sent: 19 October 2004 16:18
  To: [EMAIL PROTECTED]
  Subject: [Asterisk-Users] Almost there--Remote connection
  
 
 [snip]
 
  
  The * server is behind a firewall and I have opened ports
 1-10100
  5060
  5004
  4569
  
 
 IIRC, SIP uses 1-2 by default. Have you changed this to 
 1-10100?
 
 Cheers,
 
 Karl
 
 __
 __
 This e-mail has been scanned for all viruses by Star. The service is 
 powered by MessageLabs. For more information on a proactive anti-virus

 service working around the clock, around the globe, visit:
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 __
 This e-mail has been scanned for all viruses by Star. The service is 
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 service working around the clock, around the globe, visit:
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RE: [Asterisk-Users] Almost there--Remote connection

Thanks. The server is NAT'd.
So, Am I to conclude that it is not going to work and I should abandon
it?




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
on Asterisk Mailing Lists
Sent: Tuesday, October 19, 2004 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Almost there--Remote connection


On Tue, 19 Oct 2004 11:18:17 -0400, Ferguson, Michael
[EMAIL PROTECTED] wrote:
 
 My Grandstream 100 is at a remote location on broadband and connects 
 to my * server else where.

and:

 The * server is behind a firewall

and:

 The GS100 rings. When answered I can clearly hear everything coming 
 from the phone that's calling in. The caller cannot hear anything 
 coming from the GS100 IP phone.

Of course not.

Running a SIP server behind a Firewall does not exactly make things
straightforward.

Is your server is only behind a firewall or is it also behind a NAT?

If it is behind NAT you should know that that SIP/NAT traversal
workarounds are for clients behind NAT connecting to servers on public
IPs, not for clients on public IPs connecting to servers behind NAT.

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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RE: [Asterisk-Users] Almost there--Remote connection

Thanks.
I opened 1-2 also on the remote firewall, but still no success.
Quite frustrating.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl Dyson
Sent: Tuesday, October 19, 2004 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connection

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ferguson, Michael
 Sent: 19 October 2004 18:30
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Almost there--Remote connection

 I just realised that I neglected to mention that the remote
 GS100 phone is sitting behind a firewall also.
 Do I need to open any outgoing ports on that firewall? 
 Considering that one cannot hear anything from the GS100 IP phone?

Yes, both phones will need to have ports 1-2 open (having seen
your rtp.conf) if they are going o register with your * server.

 Mine says
 rtpstart=1
 rtpend=2

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RE: [Asterisk-Users] Vonage with Nat - Working

Jared,
Congrats on your success. I am still battling with mine, achieving one
way success.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jared
Watkins
Sent: Tuesday, October 19, 2004 2:19 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Vonage with Nat - Working

After much research and trial and error I've gotten a vonage softphone 
account working through a NAT firewall...  I'll be updating the wiki 
with this info after I see what sort of feedback I get from the list.  I

don't know that this is the only way to make it work.. but this way does

work for me.

I'll start with the NAT setup...it shouldn't matter.. but fyi I'm 
using a linux firewall with iptables and the fwbuilder package to create

the rules.

The following UDP ports are allowed in.. and forwarded to your internal 
asterisk box.

8000 - 8020
5060 - 5061
1 - 2

in sip.conf

[general]
externip = your external address  (possible dns name too?)

; While not required... I found the following useragent string from the 
softphone they provide... 
; it does not seem to make a difference if you use it.

;useragent = X-PRO Vonage release 1102t

; example
; register = 1704555:[EMAIL PROTECTED]:5061/vonage-in

register = your full softphone number including 1:case sensitive 
password@sphone.vopr.vonage.net:5061/incoming call context

; Next.. the specific vonage entry in sip.conf

[sphone.vopr.vonage.net]
secret = your password
username = full softphone number
insecure = very
disallow = all
allow = ulaw
port = 5061
host = sphone.vopr.vonage.net
nat = yes
type = peer
canreinvite = no
dtmfmode = rfc2833
fromuser = your softphone number
context = vonage-in

Then... in extensions.conf I have the following example for incoming 
calls

[vonage-in]
exten = 1NXXNXXX,1,-- some acton

For outgoing calls I use something like this...

Dial(SIP/[EMAIL PROTECTED])

I use macros in my extensions file... but that's the basic idea...   
With these settings I'm able to make and receive calls using a vonage 
softphone account from behind a NAT firewall.  I hope that is of some 
use to others out there...

Jared

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RE: [Asterisk-Users] Almost there--Remote connection

Ryan,
Thanks. That looks hopeful.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan
Courtnage
Sent: Tuesday, October 19, 2004 2:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connection


On Tue, 2004-19-10 at 14:07 -0400, Ferguson, Michael wrote:
 Thanks. The server is NAT'd.
 So, Am I to conclude that it is not going to work and I should abandon

 it?

I've been down this road.

Follow this thread:
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/45339

Ryan Courtnage

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RE: [Asterisk-Users] Almost there--Remote connection