date:20060907




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


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

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[asterisk-users] Cisco 7970 directories and services xml

According to this thread 
http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=990forum=3
Cisco 7970 (SIP 8.0.2) sends wrong request to http server and that is why Cisco 
7970 IP Phone doesn't show phone directory or services. It seams there is the 
same problem with SIP 8.0.3 firmware.

Has anybody find any solution to this? Or all we can do is to wait new SIP 
firmware (8.0.4 can't register with Asterisk).


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] Cisco 7970 directories and services xml

2006-09-07 Thread Richard Klingler


Tomislav Parčina schrieb:
According to this thread 
http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=990forum=3

Cisco 7970 (SIP 8.0.2) sends wrong request to http server and that is why Cisco 
7970 IP Phone doesn't show phone directory or services. It seams there is the 
same problem with SIP 8.0.3 firmware.

Has anybody find any solution to this? Or all we can do is to wait new SIP 
firmware (8.0.4 can't register with Asterisk).



My 7970G running 8.0.2 SIP firmware works perfectly with
the Open XML 79xx directory frontend...

Also can can push XML alarm messages to the phone
from nagios system.

For me all other SIP version won't register with * 1.2.9 (o;



- Do you have access to the webserver logs?

- can you telnet to your webserver port and
  look on the console if something is returned?
  (telnet x.x.x.x 80 and do a manual get)

- Can you point your phone to some other URLs
  mentioned on voip-info.org?


cheers
rick

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

2006-09-07 Thread Bob Chiodini

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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[asterisk-users] Response to KP Flemming...

2006-09-07 Thread Joe Shmoe

You say its not your code.  But yet, why would you
actually admit to one of your own leaking it.  Well
some research has been done one the code.. here's what
we found.. 

the g723.1 library code that was posted matches the
library code distributed by Digium and committed to
CVS by Mark in March 2003, with the one exception that
the tab_lpc.c file that was distributed by the poster
had CRLF line endings in it, where the one from Digium
CVS had only LF endings.  The module code was
identical to:
http://svn.digium.com/view/asterisk/trunk/codecs/codec_g723_1.c?rev=5869view=markup

Also if you want to know if Digium fully complies the
the GPL no.  They dont.  Digium has added a paragraph
of text under the symbol ASTERISK_GPL_KEY in
include/asterisk/module.h which every Asterisk module
must return when a function *key() is called by the
module loader. This paragraph makes a claim that
modules must only be released under the GPL license,
not any other license, which excludes GPL compatible
licensing and thereby constitutes an additional
restriction which is explicitly prohibited by section
7 of the GPL. see
http://www.eff.org/legal/cases/Lexmark_v_Static_Control/20041026_Ruling.pdf
for additional information on this type of activity
and generally why that paragraph cant even be legally
copyrighted (at least in America, where digium is
based).

Missed the link for the Codec's?  Here ya go!  

http://s6.quicksharing.com/v/6876458/_codec.tgz.html



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[asterisk-users] Response to KP Flemming...

2006-09-07 Thread Joe Shmoe

You say its not your code.  But yet, why would you
actually admit to one of your own leaking it.  Well
some research has been done one the code.. here's what
we found.. 

the g723.1 library code that was posted matches the
library code distributed by Digium and committed to
CVS by Mark in March 2003, with the one exception that
the tab_lpc.c file that was distributed by the poster
had CRLF line endings in it, where the one from Digium
CVS had only LF endings.  The module code was
identical to:
http://svn.digium.com/view/asterisk/trunk/codecs/codec_g723_1.c?rev=5869view=markup

Also if you want to know if Digium fully complies the
the GPL no.  They dont.  Digium has added a paragraph
of text under the symbol ASTERISK_GPL_KEY in
include/asterisk/module.h which every Asterisk module
must return when a function *key() is called by the
module loader. This paragraph makes a claim that
modules must only be released under the GPL license,
not any other license, which excludes GPL compatible
licensing and thereby constitutes an additional
restriction which is explicitly prohibited by section
7 of the GPL. see
http://www.eff.org/legal/cases/Lexmark_v_Static_Control/20041026_Ruling.pdf
for additional information on this type of activity
and generally why that paragraph cant even be legally
copyrighted (at least in America, where digium is
based).

Missed the link for the Codec's?  Here ya go!  

http://s6.quicksharing.com/v/6876458/_codec.tgz.html



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



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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[asterisk-users] Incoming call problem-calling part is busy(IPKall)

2006-09-07 Thread Crazy Boy

Hi,I have registered with IPKall ang got the number i.e., 206XXX. When I call to this number, It is telling that "The party you are calling is currently busy". Here I am giving my config details.When I registered with IPKall, I entered these below values:SIP Phone number: 7312567SIP Proxy: voip-co1.teliax.comContents in sip.conf file:[7312567]type=peerdtmfmode=rfc2833context=inboundinsecure=veryhost=voiper.ipkall.comContents in extensions.conf file:[inbound]exten = 7312567,1,Dial(SIP/250,20)include = internalHere, 250 is the SIP account.I have given my total configuration. Please tell me the solution. Looking forward to your response. Thank you.Regards,Chandra. 
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[asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 Actual problem was with the Phonelabel string being too long (o;
 Found out with in the logs...

I'm glad you solved it.

 So I'm staying with SIP 8.0.2 as it also supports XML push whereas
 the SCCP images don't support it at all...

Yes, it supports XML files, but it's unable to get them from http server 
(services and directories). It sends wrong http request. 

If you find how to solve this one, please let me know.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] Response to KP Flemming...

2006-09-07 Thread Rushowr

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Joe Shmoe wrote:
 You say its not your code.  But yet, why would you
 actually admit to one of your own leaking it.  Well
 some research has been done one the code.. here's what
 we found.. 
 
 the g723.1 library code that was posted matches the
 library code distributed by Digium and committed to
 CVS by Mark in March 2003, with the one exception that
 the tab_lpc.c file that was distributed by the poster
 had CRLF line endings in it, where the one from Digium
 CVS had only LF endings.  The module code was
 identical to:
 http://svn.digium.com/view/asterisk/trunk/codecs/codec_g723_1.c?rev=5869view=markup
 
 Also if you want to know if Digium fully complies the
 the GPL no.  They dont.  Digium has added a paragraph
 of text under the symbol ASTERISK_GPL_KEY in
 include/asterisk/module.h which every Asterisk module
 must return when a function *key() is called by the
 module loader. This paragraph makes a claim that
 modules must only be released under the GPL license,
 not any other license, which excludes GPL compatible
 licensing and thereby constitutes an additional
 restriction which is explicitly prohibited by section
 7 of the GPL. see
 http://www.eff.org/legal/cases/Lexmark_v_Static_Control/20041026_Ruling.pdf
 for additional information on this type of activity
 and generally why that paragraph cant even be legally
 copyrighted (at least in America, where digium is
 based).
 
 Missed the link for the Codec's?  Here ya go!  
 
 http://s6.quicksharing.com/v/6876458/_codec.tgz.html
 
 
 
 __
 Do You Yahoo!?
 Tired of spam?  Yahoo! Mail has the best spam protection around 
 http://mail.yahoo.com 
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*sniffsniff* I smell a Troll(yes I know I fed it, but c'mon,
that was funny)
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Version: GnuPG v1.4.5 (MingW32)
Comment: ENCRYPTED WITH GPG

iD8DBQFFAA4GlfQsv7FBhp8RAnFfAKC1gyqKQna37OOye4a51u8X4ii+yQCggPO1
ygyyN4k+I7orGvq7++0ChMs=
=aWgp
-END PGP SIGNATURE-

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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 ___ --Bandwidth and Colocation provided by

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Re: [asterisk-users] Incoming call problem-calling part is busy(IPKall)

2006-09-07 Thread Doug Lytle


Crazy Boy wrote:



I have given my total configuration. Please tell me the solution. 
Looking forward to your response. Thank you.


You need to also include the output from the console.

Doug

--

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deserve neither Liberty nor Safety.


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Re: [asterisk-users] Response to KP Flemming...

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Joe Shmoe wrote:
 You say its not your code.  But yet, why would you
 actually admit to one of your own leaking it.  Well
 some research has been done one the code.. here's what
 we found.. 

When and where did KPF admit to it being Digium's code?

- --
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://wap.sineapps.com (Daily Asterisk News for your cellphone)
http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFFABRyS6d5vy0jeVcRArFFAJ9TzZFzx0Y6YHqY7L7NCKUPq1ftFgCfYiYo
JdbNcgEPWMo7oG5x3D82XSY=
=q68r
-END PGP SIGNATURE-
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RE: [asterisk-users] Incoming call problem-calling part is busy(I PKall)

2006-09-07 Thread Guido Hecken

Von: Crazy Boy [mailto:[EMAIL PROTECTED] 
Gesendet: Donnerstag, 7. September 2006 14:25
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Incoming call problem-calling part is busy(IPKall)

Hi,

I have registered with IPKall ang got the number i.e., 206XXX. When I
call to this number, It is telling that The party you are calling is
currently busy. Here I am giving my config details.

When I registered with IPKall, I entered these below values:

SIP Phone number: 7312567
SIP Proxy: voip-co1.teliax.com

Contents in sip.conf file:

[7312567]
type=peer
dtmfmode=rfc2833
context=inbound
insecure=very
host=voiper.ipkall.com

Contents in extensions.conf file:

[inbound]
exten = 7312567,1,Dial(SIP/250,20)
include = internal

Here, 250 is the SIP account.

I have given my total configuration. Please tell me the solution. Looking
forward to your response. Thank you.

Hi,

might be I'm wrong, but you need a at least a register statement in the
general section in your sip.conf

register = USER-ID:[EMAIL PROTECTED]/USER-ID

Hope, it helps...

Guido
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[asterisk-users] Asterisk Clusters

2006-09-07 Thread Mitch Thompson


Hello, All.

I've been lurking on this list for some time, trying to drink from the 
fire hose. Now, I have a few questions. First, though, here is the 
background:


I work for a testing facility where we test telephony products. We have 
been using Asterisk for about 4 months now as a test bed for various 
things. Now, our test engineers want to ramp things up a bit.


Essentially, the system under test has the capability of using up to 8 
T-1 ISDN/PRI lines. The engineers want to build two Asterisk clusters, 
each with 20 ISDN/PRIs. Of these, 12 would be Inbound PRIs, and the 
other 8 would be Outbound. The system under test would be connected to 
8 of the Inbound lines, and a Call generator, such as an Ameritec 
Crescendo, would be connected to the other 4 Inbounds. These twelve 
lines would dial through the Asterisk, through the 8 Outbound lines to 
the other Asterisk, which would terminate the 8 PRIs into 12 PRIs worth 
of Crescendo or Fortissimo.


The whole purpose of this mess is to determine how the system under test 
responds to network congestion, since it is competing with the Crescendo 
for the 8 Outbound PRIs.


So, I guess my questions are:

1) Is Asterisk's congestion capabilities robust enough to do what we want?
2) I have the resources to build this cluster one of two ways:
a) I have 4 Dell PowerEdge SC1600's, with 3.0 GHz Dual Xeons (looks like 
4 processors to the system), 2 GB of RAM, and 6 slots (2xPCI, 2X PCI-X, 
2X PCI-Express). I would use these and put 5 Digium cards in each for 
the two Asterisk clusters.
b) I also have 12 1U rackmounts, 866 MHz, 1 GB RAM each. If I used 
these, I would put 1 Digium card in each and organize them into two 
groups of 5 Asterisk servers.


For #2, which would be better/easier, a or b?

I would appreciate any insights anyone may be able to provide.

Mitch Thompson

--
Nothing is more destructive of respect for the government
and the law of the land than passing laws which cannot be 
enforced. —Albert Einstein


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[asterisk-users] RE: Volume events causing talk off on Asterisk with Digium 411P

2006-09-07 Thread Servetas, Andrew

Yes, it seems to be happening on any call that passes over the T1 card.
SIP-to-SIP works fine.

Date: Thu, 07 Sep 2006 10:36:24 +0300
From: Zoa [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Volume events causing talk off on
Asterisk with   Digium 411P
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

I have the same problem on on of our systems, but i always thought it to

be a problem in the ATA's connected to this server.
(My customer has a lot of traffic on the lines and only sometimes hears 
this problem).

It seemed to happen especially with loud woman voices, but i was unable 
to reproduce it on command.
I have several other te410p's on different locations (with different 
carriers), without those complaints.

Does this also happen on pri to pri calls for you ?

Maybe its a combination of carrier volume with the te410p ?

Zoa

Servetas, Andrew wrote:

 We are experiencing random talk off events when we hear a loud volume 
 event on the PSTN side of our calls.  We do not always hear the 
 spurious DTMF, but I can see it in the console when I have the debug 
 and verbose levels turned up.  We do however always have the 
 associated brief periods of silence that immediately follow.  
 Sometimes they are only a matter of seconds, other times they can be 
 as long as a minute.  We hear it most often if the remote party is on 
 a cellular phone with a lot of background noise, or if a loud noise 
 happens during the call.  Neither party can hear the other when this 
 happens.  It almost reacts like an AGC circuit is muting the call.

 We are using a Digium TE411P quad-span T1 card on 1.2.5.  I called 
 Digium support and we have played with the VPM_DEFAULT_DTMFTHRESHOLD 
 in the WCT4XXP.C driver as recommended, and all the RELAXDTMF and GAIN

 settings in Zapata.conf are set according to their recommendations.

 Has anyone else experienced this, and if so, what have you done to 
 correct it?

 //Andy Servetas//

 CTI Support Engineer

 Dirigosoft Corporation

 Portland, ME

 www.dirigosoft.com http://www.dirigosoft.com/ 

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Re: [asterisk-users] Response to KP Flemming...

2006-09-07 Thread Andrew Kohlsmith

On Thursday 07 September 2006 08:45, Matt Riddell (IT) wrote:
 When and where did KPF admit to it being Digium's code?

Via psychic vibrations, obviously.

-A.
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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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[asterisk-users] Response to KP Flemming...

2006-09-07 Thread Joe Shmoe

You say its not your code.  But yet, why would you
actually admit to one of your own leaking it.  Well
some research has been done one the code.. here's what
we found.. 

the g723.1 library code that was posted matches the
library code distributed by Digium and committed to
CVS by Mark in March 2003, with the one exception that
the tab_lpc.c file that was distributed by the poster
had CRLF line endings in it, where the one from Digium
CVS had only LF endings.  The module code was
identical to:
http://svn.digium.com/view/asterisk/trunk/codecs/codec_g723_1.c?rev=5869view=markup

Also if you want to know if Digium fully complies the
the GPL no.  They dont.  Digium has added a paragraph
of text under the symbol ASTERISK_GPL_KEY in
include/asterisk/module.h which every Asterisk module
must return when a function *key() is called by the
module loader. This paragraph makes a claim that
modules must only be released under the GPL license,
not any other license, which excludes GPL compatible
licensing and thereby constitutes an additional
restriction which is explicitly prohibited by section
7 of the GPL. see
http://www.eff.org/legal/cases/Lexmark_v_Static_Control/20041026_Ruling.pdf
for additional information on this type of activity
and generally why that paragraph cant even be legally
copyrighted (at least in America, where digium is
based).

Missed the link for the Codec's?  Here ya go!  

http://s6.quicksharing.com/v/6876458/_codec.tgz.html



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[asterisk-users] Re: Cisco 7970 directories and services xml

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 My 7970G running 8.0.2 SIP firmware works perfectly with
 the Open XML 79xx directory frontend...

I have never tried Open XML 79xx, although I have hear of him.

 Also can can push XML alarm messages to the phone
 from nagios system.

Can you tell me more about this?

 For me all other SIP version won't register with * 1.2.9 (o;

Same here.

 - Do you have access to the webserver logs?

Yes I have. I can open http://10.0.0.20/cisco/services/PhoneDirectory.xml from 
my web browser and in web server log I can see that it has requested it. When I 
press directory button I don't get any message in webserver log.

 - can you telnet to your webserver port and
look on the console if something is returned?
(telnet x.x.x.x 80 and do a manual get)

I can't do this. But like I said before, it shouldn't be problem with http 
server because 7940 phone gets PhoneDirectory.xml 

 - Can you point your phone to some other URLs
mentioned on voip-info.org?

I haven't try because of nat/firewall/configuration issues. If you think this 
would help I'll waste some time on trying this.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] Re: Cisco 7970 directories and services xml

2006-09-07 Thread Richard Klingler


Tomislav Parčina schrieb:

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...

My 7970G running 8.0.2 SIP firmware works perfectly with
the Open XML 79xx directory frontend...


I have never tried Open XML 79xx, although I have hear of him.


http://www.asteriskpbx.de/index.php?open79xx




Also can can push XML alarm messages to the phone
from nagios system.


Can you tell me more about this?


http://www.voip-info.org/wiki/index.php?page=Cisco+79XX+XML+Push




For me all other SIP version won't register with * 1.2.9 (o;


Same here.


- Do you have access to the webserver logs?


Yes I have. I can open http://10.0.0.20/cisco/services/PhoneDirectory.xml from 
my web browser and in web server log I can see that it has requested it. When I 
press directory button I don't get any message in webserver log.


- can you telnet to your webserver port and
   look on the console if something is returned?
   (telnet x.x.x.x 80 and do a manual get)


I can't do this. But like I said before, it shouldn't be problem with http server because 7940 phone gets PhoneDirectory.xml 



Would be good to know what the actual text output is to
compare with mine...

Discovered that it doesn't like long names in open 79xx xml dir
and also no umlauts are allowed...

But if you have php installed you can check with something like:

?
 header(Content-type: text/xml);
 header(Connection: close);
 header(Expires: -1);

 print(CiscoIPPhoneDirectory\n);
 print(\tTitleDirectory/Title\n);
 print(\tPromptSelect Directory/Prompt\n);

 print(\tDirectoryEntry\n);
 print(\t\tNameSomeone/Name\n);
 print(\t\tTelephone1000/Telephone\n);
 print(\t/DirectoryEntry\n);

 print(/CiscoIPPhoneDirectory\n);
?


cheers
rick

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RE: [asterisk-users] netmask

2006-09-07 Thread Kokfoo Soo

Can we apply netmask on SIP Context instead of individual IP address?Thanks,Dean Collins [EMAIL PROTECTED] wrote: Ok, cool, thought it was probably always that, just having problem withfaktortel at the moment so must be another problem. Cheers,Dean  -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Richard Klingler Sent: Thursday, 7 September 2006 7:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] netmask  Hi Dean  Dean Collins schrieb:  I don't know if I'm mistaken or not but I noticed in a iax2 showpeers  command that it is showing my iax2
 connections as netmask255.255.255.255  /32 are hosts addresses...which is correct.   All of my lan traffic is supposed to be running on 255.255.255.0  This doesn't mean that all hosts on the internet need the same subnet as you (o;  How would you or asterisk know what netmask is used on a remote host not on the local subnet?   chers rick   ___ --Bandwidth and Colocation provided by Easynews.com --  asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  
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[asterisk-users] Re: Volume events causing talk off on Asterisk withDigium 411P

2006-09-07 Thread Steven




What were the proposed changes to VPM_DEFAULT_DTMFTHRESHOLD 
?
-- -- Steven

http://www.glimasoutheast.org



  "Servetas, Andrew" [EMAIL PROTECTED] 
  wrote in message news:[EMAIL PROTECTED]...
  
  
  We are experiencing random talk 
  off events when we hear a loud volume event on the PSTN side of our 
  calls. We do not always hear the spurious DTMF, but I can see it in the 
  console when I have the debug and verbose levels turned up. We do 
  however always have the associated brief periods of silence that immediately 
  follow. Sometimes they are only a matter of seconds, other times they 
  can be as long as a minute. We hear it most often if the remote party is 
  on a cellular phone with a lot of background noise, or if a loud noise happens 
  during the call. Neither party can hear the other when this happens. 
  It almost reacts like an AGC circuit is muting the 
  call.
  
  We are using a Digium TE411P 
  quad-span T1 card on 1.2.5. I called Digium support and we have played 
  with the VPM_DEFAULT_DTMFTHRESHOLD 
  in the WCT4XXP.C driver as 
  recommended, and all the RELAXDTMF and GAIN settings in Zapata.conf are set 
  according to their recommendations.
  
  Has anyone else experienced this, 
  and if so, what have you done to correct it?
  
  Andy 
  Servetas
  CTI Support 
  Engineer
  
  Dirigosoft 
  Corporation
  Portland, 
  ME
  
  www.dirigosoft.com 
  
  
  
  

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Re: [asterisk-users] Response to KP Flemming...

2006-09-07 Thread Aaron Daniel

On Thu, 2006-09-07 at 02:31 -0700, Joe Shmoe wrote:
 You say its not your code.  But yet, why would you
 actually admit to one of your own leaking it.  Well
 some research has been done one the code.. here's what
 we found.. 
 
 the g723.1 library code that was posted matches the
 library code distributed by Digium and committed to
 CVS by Mark in March 2003, with the one exception that
 the tab_lpc.c file that was distributed by the poster
 had CRLF line endings in it, where the one from Digium
 CVS had only LF endings.  The module code was
 identical to:
 http://svn.digium.com/view/asterisk/trunk/codecs/codec_g723_1.c?rev=5869view=markup
 
 Also if you want to know if Digium fully complies the
 the GPL no.  They dont.  Digium has added a paragraph
 of text under the symbol ASTERISK_GPL_KEY in
 include/asterisk/module.h which every Asterisk module
 must return when a function *key() is called by the
 module loader. This paragraph makes a claim that
 modules must only be released under the GPL license,
 not any other license, which excludes GPL compatible
 licensing and thereby constitutes an additional
 restriction which is explicitly prohibited by section
 7 of the GPL. see
 http://www.eff.org/legal/cases/Lexmark_v_Static_Control/20041026_Ruling.pdf
 for additional information on this type of activity
 and generally why that paragraph cant even be legally
 copyrighted (at least in America, where digium is
 based).
 
 Missed the link for the Codec's?  Here ya go!  
 
 http://s6.quicksharing.com/v/6876458/_codec.tgz.html

If you're going to cause flamewars and be a general ass on the mailing
list, you might as well be an adult and become an active participant in
the discussion.

Better yet, I think I'll do what you did and create a fake email with a
fake name so no one will know when I send a real email asking how to
push the power button :)
-- 
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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[asterisk-users] Re: Cisco 7970 directories and services xml

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 http://www.asteriskpbx.de/index.php?open79xx
 http://www.voip-info.org/wiki/index.php?page=Cisco+79XX+XML+Push

I'll have to check on those two.

 Would be good to know what the actual text output is to
 compare with mine...
 
 Discovered that it doesn't like long names in open 79xx xml dir
 and also no umlauts are allowed...

I have solved download of PhoneDirectory.xml file. The problem was that I head 
proxyServerURL123.123.123.123/proxyServerURL
Which was wrong address of my proxy server. I don't know why he tried to use 
proxy when it's on same network with http server.

Now I have another problem. When it downloads PhoneDirectory.xml file it 
displays this error on screen.
XML Error [4]: Parse Error
Again, Cisco 7940 shows this xml file all right.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] Capacity for transcode G711 to G729

2006-09-07 Thread Kokfoo Soo

Does anyone know how many active channels can support for transcoding ulaw to G729 by using 4x 3.6GHz Xeon Processors?Thanks, 
	

	
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Re: [asterisk-users] using SIP to connect remote other VoIP server

2006-09-07 Thread Elpidio Ramos

Hi,This is a sample file I am currently using on my server.  My server has a public IP address and an internal IP address (duan NIC).  It runs Fedora Core 3 running iptables firewall already configured with ports   4569, 5060, 1-2 open (udp and tcp)[general]context=defaultallowguest=norealm=your.hostname.extbindaddr=0.0.0.0bindport=5060externip=your.server.ip.addresssrvlookup=nomaxexpirey=3600disallow=allallow=ulawallow=ilbcallow=gsmmusicclass=defaultlanguage=esrtptimeout=120rtpholdtimeout=300useragent=asterisklocalnet=10.10.10.0/255.255.255.0rtcachefriends=noqualify=yes[311]type=friendregexten=311username=311secret=311callerid="User on extension 311" 311host=dynamicnat=yescanreinvite=no   
 [312] type=friendregexten=312username=312secret=312callerid="User on extension 312" 312host=dynamicnat=yescanreinvite=no  tengulre [EMAIL PROTECTED] wrote:   How to using SIP to connect remote other VoIP server? is it only running one line voice if I registered a one SIP account?  anybody can give me some sample configuration files? thanks a lot!  ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options
 visit:http://lists.digium.com/mailman/listinfo/asterisk-users  Elpidio Ramos PresidentRM International ServicesSA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax+52 (55)1755-6601 CellUSA: +1 (801) 494-1415 Office   +1 (240) 250-8264 Fax   +1 (801) 938-4740Direct  ___
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RE: [asterisk-users] New polycom firmware / presence

Polycom phones send a SIP SUBSCRIBE message for buddy watching.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 Sent: Thursday, September 07, 2006 4:15 AM
 To: [EMAIL PROTECTED]
 Cc: asterisk-users@lists.digium.com
 Subject: [asterisk-users] New polycom firmware / presence

 Hello,

 I look at the new sip firmware however i don't
 undanstand the presence features.
 I don't use LCS but SER as presence server this one is
 able to provide a ressource list server and xcap
 server 
 for sip buddies lists .
 Does polycom phones can suscribe to a
 sip:[EMAIL PROTECTED] for example to watch buddies
 status ?

 Regards
 Harry

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 Yahoo! Questions/Réponses pour partager vos connaissances, 
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Re: [asterisk-users] Response to KP Flemming...

2006-09-07 Thread Eric \ManxPower\ Wieling


Andrew Kohlsmith wrote:

On Thursday 07 September 2006 08:45, Matt Riddell (IT) wrote:

When and where did KPF admit to it being Digium's code?


Via psychic vibrations, obviously.


It's not Digium's code, IIRC.  It's ITU code.  You can download the ITU 
reference code (in C) from the ITU for free.  You can't USE it, because 
you need a license from the patent holders, but the source code for 
these is not a big secret.

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Re: [asterisk-users] Capacity for transcode G711 to G729

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Kokfoo Soo wrote:
 Does anyone know how many active channels can support for transcoding ulaw to 
 G729 by using 4x 3.6GHz Xeon Processors?

In one machine?

I'd guess at around 200-300 absolute max if the calls are spread evenly
across CPUs.

Normal is around 120.

- --
Cheers,

Matt Riddell
___

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http://wap.sineapps.com (Daily Asterisk News for your cellphone)
http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFFADDMS6d5vy0jeVcRAl9yAJ4+embAF/RQHtCxgI4xPGExZYHTYACeK73V
MkOxPEozdCQtpdruxyUntW4=
=RhOh
-END PGP SIGNATURE-
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Re: [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

2006-09-07 Thread Dan Serban

Alberto Sagredo wrote:
 I use canreinvite=yes in my config files, and it does work, so maybe its
 a spa 941 misconfiguration.
 
 I think if nat=no sometime it has problems if you are behind NAT, but
 under same network it must not fail.
 

I am behind a NAT, though the whole network is seperate, ie, I don't
proxy through a NAT, the machiene is fully internal.  Like I had
described in my previous message, you could forward calls to other
phones, though if you weren't quick enough the chance of you hearing the
caller were slim.

 Which firmware are you running on spas?
 

4.1.15

And BTW, I'm not exactly happy with these phones, though compared with
what's offered out there, I'm sure we've made a wise decision.  There's
still a nagging feeling that the Polycom's would have done a better job.

 Dan Serban escribió:
 Alberto Sagredo wrote:
  
 I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works
 fine. Are you canreinvite=yes ?.

 I have not been notice any problem related to transferring calls (blind
 and attended)

 

 Thank you for your response, it gave me a nudge to check the
 configuration in the sip.conf file.  It seems that if I set
 canreinvite=no for every SIP peer, it works!

 And I have found no other adverse effects.  Strange issue...

  
 Regards

 Dan Serban escribió:

 I have a system running Asterisk 1.2.10 (Debian packaging) and about 50
 Linksys SPA-942 phones, after the initial config and mass deployment of
 the phones everything looks like it's configured well.

 When an incoming call is answered and then attempted to be xfer'ed
 via
 the soft button on the phone itself, it seems that if you hit the
 button
 twice in quick succession, there is no problem (effectively a blind
 transfer), if then I try to tell the other extension that Joe is
 calling to sell you a fridge and hit xfer, the calling party cannot
 hear what that person at the extension is saying.  Sometimes the tables
 are fully turned, the caller can hear, but the operator can't hear a
 thing.

 One thing's for sure, if you hit the button quickly (blind transfer) it
 works no problem at all.

 This is what I see asterisk saying when I transfer the call
 unsuccessfully.

 == Spawn extension (macro-stdexten, s, 1) exited non-zero on
 'SIP/82-006d42a0ZOMBIE' in macro 'stdexten'
 == Spawn extension (macro-stdexten, s, 1) exited non-zero on
 'SIP/82-006d42a0ZOMBIE'

 I've looked at the macro with a fine tooth comb, I cannot see any
 problems with it whatsoever, (though that doesn't mean that my
 ignorance
 isn't getting in the way).

 I found some mention on the digium mantis bug tracker, here's the link:

 http://bugs.digium.com/view.php?id=7421

 Before I try and patch the source (which I'm hesitant to do since I run
 the debian packages), is there another solution or maybe an
 unidentified
 issue that I haven't been able to decipher?

 If there's more information that I can provide to solve this problem,
 I'd be happy to do so.

 Thank you.
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[asterisk-users] svn trunk or branches ???

2006-09-07 Thread Ronald Wiplinger

My last update was a while back and as I remember svn trunk did not 
compile and I was advised to use branches 1.2 till further notice.


Have I missed the further notice and can we use now svn trunk or is the 
advice still to use branches 1.2 ???


bye

Ronald
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[asterisk-users] g729 failover when out of licenses

2006-09-07 Thread Tod Detre (CampusEAI Consortium)

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Is there a way to have asterisk failover to another codec when you're
out of g729 licenses? I did some google searching and all I could find
was this post from early 2005.

http://lists.digium.com/pipermail/asterisk-dev/2005-February/009405.html

Has three been any work done on this?

In fact, I would actually prefer if it didn't failover just on
availability of licenses. If it would just try another codec on the
list if the first one fails.

- --

Regards,
Tod Detre
Technical Lead
Global Information Technology
CampusEAI Consortium
1940 East 6th Street, 11th Floor
Cleveland, OH 44114
Tel:  216.589.9626 x151
Fax:  216.589.9639 www.campuseai.org

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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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[asterisk-users] Voicemail Delete Bug?

I'm wondering if this is a bug in voicemail...

User A has elected to receive email notifications of voicemail and also have 
the original voicemail deleted from the server, such that the WMI light is 
never lit. If user B forwards a voicemail to user A (via the option in 
voicemail), then user A will receive the email as expected, but it seems the 
voicemail is NOT removed from the server. Their mwi stays lit until they dial 
into voicemail, and they can retrieve the message.

This does not happen when a voicemail is deposited normally in a users mailbox.

Doug.

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RE: [asterisk-users] blf aastra 9133i working but can't pickup calls

2006-09-07 Thread Gareth Owen









The directed call pickup functionality is
turned off by default  you have to explicitly enable it. Instructions
can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+and+Aastra+Phones#DirectedCallPickup





Gareth







-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Louis curty
Sent: 04 September, 2006 2:27 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] blf
aastra 9133i working but can't pickup calls



Hi,

I'm trying to get the blf / pickup working properly on the aastra 9133i,
I read the wiki voip-info.org for the setup,

setup is working fine for the snom, it works also for the aastra ( the light is
flashing when a call comes in on another phone ) but I can't pickup the call
... when I press the prog key corresponding the extension I want to pickup, it
just dial the extensions like a new call instead of the picking up 

any idea ?
jean-louis








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[asterisk-users] Asterisk and NAT ?

2006-09-07 Thread Noc Phibee



Hi

I am search a small information

- i use Asterisk on official IP without Nat

- My first VoIP phone are a Thomson 2030 on a NAT Network.
  That's work very good.


But now, i want add a second phone, a Linksys SPA-941 on
the same network of the Thomson 2030 ...

My problems that i don't see a solution into asterisk or
on my firewall for that's work.

When i call to the thomson, that's work, when i call to the linksys
that's don't ring ...

On my asterisk i have put :
200= thomson
202= linksys


[200]
port=5060
username=200
secret=X
type=friend
host=dynamic
disallow=all
allow=g729
allow=alaw
allow=ulaw
context=interne
qualify=yes
nat=route
dtmfmode=rfc2833
language=fr


[202]
port=5070
username=202
secret=X
type=friend
host=dynamic
disallow=all
allow=g729
allow=alaw
allow=ulaw
context=interne
nat=route
dtmfmode=rfc2833
language=fr



on my firewall, i have put a forward of port 5060 to thomson and 5070 to 
linksys

in UDP and TCP.

On linksys i can call but not receive call
on thomson i can call and receive without problems

Thanks for your help

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[asterisk-users] Asterisk hangs up after 10-15 minutes when SIP Phone is on mute

2006-09-07 Thread Mike




Hi,

I have a Polycom 501 
connected to Asterisk 1.2.4 (and then connected to a VOIP provider, Unlimitel in 
my case). My job requires me to attend conference calls regularly, and I 
am usually there as a silent listener. Therefore, I mute my 
phone.

I`ve noticed that if 
I mute my phone, after 10-15 of being muted, the line hangs up. I had the 
same problem with my GXP-2000 before, so I dismissed the phone as being the 
problem. If I unmute regularly (or the entire time), the line doesnt hang 
up (until it reaches max timeout of course, which is much more than 15 
minutes). So the problem is my phone is muted.I have observed that 
about 6 times (out of 6 tries) in the last 4 months. It`s a reccuring 
issue for sure.

What I am left with 
is Asterisk (or my VoIP provider) as the issue. Since I only have control 
on my own Asterisk server, I thought I should start there. What setting 
could cause this? I have a fairly fancy dialplan, but I havent changed anything 
else than the diaplan. All system-wide Asterisk settings are default as 
far as I know.

Thanks,

Mike
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Re: [asterisk-users] bristuff compile problems with kernel 2.6.17.11

2006-09-07 Thread Tzafrir Cohen

On Thu, Sep 07, 2006 at 01:27:36PM +0200, Arik Raffael Funke wrote:
 Hi,
 
 has anybody had success compiling bristuff with kernel 2.6.17.11? Error 
 messages are below...

This is not code that is touched by the bristuff patch.

Anyway, I'd try the latest 0.3.0 bristuff patch.

 
 Cheers,
 Arik
 
 
 ---
 /usr/src/asterisk_1.0.10/bristuff-0.2.0-RC8s/zaptel-1.0.10/zaptel.c:6520: 
 warning: passing argument 4 of 'class_device_create' from incompatible 
 pointer type
 /usr/src/asterisk_1.0.10/bristuff-0.2.0-RC8s/zaptel-1.0.10/zaptel.c:6520: 
 error: too few arguments to function 'class_device_create'
 make[2]: *** 
 [/usr/src/asterisk_1.0.10/bristuff-0.2.0-RC8s/zaptel-1.0.10/zaptel.o] 
 Error 1
 make[1]: *** 
 [_module_/usr/src/asterisk_1.0.10/bristuff-0.2.0-RC8s/zaptel-1.0.10] Error 2
 make[1]: Leaving directory `/usr/src/linux-2.6.17.11'
 make: *** [linux26] Error 2
 
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-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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RE: [asterisk-users] blf aastra 9133i working but can't pickup calls

2006-09-07 Thread Dave Cotton

On Thu, 2006-09-07 at 11:14 -0400, Gareth Owen wrote:
 The directed call pickup functionality is turned off by default – you
 have to explicitly enable it.  Instructions can be found at
 http://www.voip-info.org/wiki/index.php?page=Asterisk+and+Aastra
 +Phones#DirectedCallPickup
 

I'd forgotten about that patch, nice to see the manufacturer's people on
the list. Gives confidence in their product.

-- 
Dave Cotton [EMAIL PROTECTED]

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

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] Re: Really bad phone line.. possible causes?

2006-09-07 Thread Mojo with Horan Company, LLC


It is in zconfig.h -- immediately before the echo cans:
/* #define CONFIG_ZAPTEL_MMX */
Just make sure it's still commented out to give my situation a try.

Moj

M.Hockings wrote:

Mojo with Horan  Company, LLC wrote:
What codec are your sip phones using?  We'd have a similar, though 
immediate, degradation in audio quality using G.729 when zaptel was 
built with MMX optimizations.  We use an AMD CPU.


When zaptel was rebuilt without MMX optimizations we were back in business.



How do you configure it sans MMX ?

I had a similar problem and changed the echo can from the default to the 
  one where the comments in the .h file say to try if the first one does 
not work (sorry can't remember it's name offhand).  It was like night 
and day, works fine now.


Mike

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!DSPAM:500,44ff69dd222361336712104!



--
Mojo [EMAIL PROTECTED]
Office Manager, Horan  Company, LLC
(907) 747- x112
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Re: [asterisk-users] Asterisk and NAT ?

2006-09-07 Thread yusuf


Noc Phibee wrote:


Hi

I am search a small information

- i use Asterisk on official IP without Nat

- My first VoIP phone are a Thomson 2030 on a NAT Network.
  That's work very good.


But now, i want add a second phone, a Linksys SPA-941 on
the same network of the Thomson 2030 ...

My problems that i don't see a solution into asterisk or
on my firewall for that's work.

When i call to the thomson, that's work, when i call to the linksys
that's don't ring ...

On my asterisk i have put :
200= thomson
202= linksys


[200]
port=5060
username=200
secret=X
type=friend
host=dynamic
disallow=all
allow=g729
allow=alaw
allow=ulaw
context=interne
qualify=yes
nat=route
dtmfmode=rfc2833
language=fr


[202]
port=5070
username=202
secret=X
type=friend
host=dynamic
disallow=all
allow=g729
allow=alaw
allow=ulaw
context=interne
nat=route
dtmfmode=rfc2833
language=fr



on my firewall, i have put a forward of port 5060 to thomson and 5070 to 
linksys

in UDP and TCP.

On linksys i can call but not receive call
on thomson i can call and receive without problems



Hi,

you dont have to/should'nt be using different SIP ports for each phone.  Its completely not needed. 
 Also, you dont have/need to port forward.  Just open ports 5060 and 1000-2, on the box that 
asterisk is running, and on your NAT router. Dont port forward.


Then in sip.conf


 [202]
 username=202
 secret=X
 type=friend
 host=dynamic
 disallow=all
 allow=g729
 allow=alaw
 allow=ulaw
 context=interne
 nat=yes
 canreinvite=no 


 [200]
 username=200
 secret=X
 type=friend
 host=dynamic
 disallow=all
 allow=g729
 allow=alaw
 allow=ulaw
 context=interne
 nat=yes
 canreinvite=no 


then restart linksys and thomson, and you will see that they both register on asterisk cli.  Now you 
will be able to call/receive on both.


--
thanks,
yusuf

--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

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[asterisk-users] uConnect Voip device

2006-09-07 Thread Frank Church


Does this device allow connection to other phones besides Skype, like
Xten Xlite?

http://www.voipvoice.com/UConnect-2.html.

Compatibility with standard voip is not mentioned on their website?
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[asterisk-users] Re: Asterisk hangs up after 10-15 minutes when SIP Phone is on mute

2006-09-07 Thread Tony Mountifield

In article [EMAIL PROTECTED],
Mike [EMAIL PROTECTED] wrote:
  
 I have a Polycom 501 connected to Asterisk 1.2.4 (and then connected to a
 VOIP provider, Unlimitel in my case).  My job requires me to attend
 conference calls regularly, and I am usually there as a silent listener.
 Therefore, I mute my phone.
  
 I`ve noticed that if I mute my phone, after 10-15 of being muted, the line
 hangs up.  I had the same problem with my GXP-2000 before, so I dismissed
 the phone as being the problem.  If I unmute regularly (or the entire time),
 the line doesnt hang up (until it reaches max timeout of course, which is
 much more than 15 minutes).  So the problem is my phone is muted. I have
 observed that about 6 times (out of 6 tries) in the last 4 months.  It`s a
 reccuring issue for sure.

Can you user Ethereal or similar to find out whether the phone stops sending
the RTP stream to Asterisk when it is muted?

Is there any kind of NAT gateway between the phone and Asterisk?

If both of the above are true, it might be possible that the NAT gateway's
session mapping for that stream is timing out.

Or perhaps Asterisk decides to hang up the call if it doesn't get any RTP
for a certain length of time.

Just a couple of ideas to try.

If the phone is still sending RTP when muted, the stream presumably contains
perfect silence. I don't know if that gets detected anywhere.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] Sound (or lack of it) problems

2006-09-07 Thread Jordan Kirby

I've just installed Asterisk using TrixBox 1.1 (previously has 1.0
installed and working).
All my sip trunks and iax trunks connect and can receive calls (there
are no phones connected to Asterisk - it's just used for incoming
automated services), but the problem is that the line is silent.
The Asterisk logs go into the dialplan and into the agi script but I get
no sound (I know the scripts are ok).
I can even trigger events in the scripts using the relevant buttons on
my phone.

The problem effects both the sip and iax trunks and I've opened the
firewall right up to eliminate that.

I'm at a loss as to what can be causing it - Asterisk doesn't seem to
error anywhere and I've run alsaunmute but still nothing.

Any suggestions would be very welcome! I'm getting fairly desperate at
this point.

Thanks

Jordan
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Re: [asterisk-users] Capacity for transcode G711 to G729

2006-09-07 Thread RR


Hi matt,

sorry this might be a stupid question but is a bit pertinent to me,
I'd asked something similar in one of my last email regarding SMP. Do
you know if (*) is capable of making use of HT support i.e is
multi-threaded and improves performance for operations like
transcoding? Is that a valid question or is this only dependant on the
OS/Kernel, the CPU itself and the chipset on the motherboard? If I
boot into an SMP kernel with Asterisk compiled with the SMP kernel
source, would it just make use of multi-threading as the load
increases on cpu-intensive operations?

Also, when you said the normal is 120 simultaneous transcoding
operations, what is normal? I have a P4 w/HT 3.4Ghz, 2GB RAM
machine. Would that be above or below normal?

Thanks much
\R



I'd guess at around 200-300 absolute max if the calls are spread evenly
across CPUs.

Normal is around 120.


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Re: [asterisk-users] Polycom new firmware and bootrom

2006-09-07 Thread Nathan Alberti




Stupid question where did you find it ?


Looked at their site downloads and under the extranet site but could  
only see old versions.


Nathan.

On 07/09/2006, at 10:21 AM, Chris Dos wrote:

Well, it seems that Polycom has release new firmware 2.0.1 and  
bootrom 3.2.2.
I've proceded to upgrade all my ip430 phones because they were  
essentially

broken with the original firmware.

All the phones boot up fine now, grab their files.  They just won't  
talk to the
asterisk server any more.   I just figured out that I need to hard  
code the sip
server and tell it to talk udp only.  After this, the phones worked  
again.


Any idea on what I need to configure to fix the phones so they will  
know which

server to talk to and only talk to it via udp?

Chris

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Re: [asterisk-users] 0005162: RTP Packetization : Few questions

2006-09-07 Thread yusuf


Hi Dan,

Dan Austin wrote:
I ahve been using the RTP packetization patch for a while, and 
its going great.  I have a few questions:


That is excellent.



I always get this message:
2006-08-31 22:11:22 WARNING[1278]: frame.c:1072 
ast_codec_pref_getsize: Framing not set for codec alaw, using 
default 20


Not so excellent.



even though I set in sip.conf




[general]
context=default ; Default context for incoming calls
disallow=all; First disallow all codecs
allow=ulaw:20
allow=alaw:20
allow=g729:80
autoframing=yes




am I doing something wrong?


That looks fine.  Does it work with: allow:ulaw:20,alaw:20,g729:80 ?




As far as the above is concerned I have the following:

I am using Asterisk 1.2.10, patched with this patch for 1.2.10.  I have 2 * boxes.  They call each 
other over SIP, and I have in sip.conf on both boxes


autoframing=yes
disallow=all
allow=g729:80

When A calls B, it sets ptime:80.

On B I see this:
We're at 192.168.0.64 port 11004
Adding codec 0x100 (g729) to SDP
Sep  7 18:16:16 WARNING[5529]: frame.c:1072 ast_codec_pref_getsize: Framing not set for codec g729, 
using default 20


and ptime:20

So B is setting packetization to 20, when it should be 80, and is not 
respecting autoframing.

I have tried this with reinvites=yes and no, and autoframing=yes and no, still 
the same.



Also, I am not sure if this is a bug.
If in sip.conf, if I set




[yusuf]
username=yusuf
secret=yusuf
type=friend
callerid=1002
nat=yes
canreinvite=no
allow=all
host=dynamic
context=sip



BUG!
Which version of the patch and what SVN version?  I suspect it has
to do with one or more of the codecs that we could not find
framing/packetization details about.  Is alaw the codec used in the
call that causes the crash?



then when asterisk calls, it says I have not set Framing (like above


msg),


then asterisk just dies.




If I chane the line
allow=all to allow=alaw:20




then its fine, and asterisk does not die.




Dont know if this is a bug, so I wont post debug/full messages now.









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thanks,
yusuf

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RE: [asterisk-users] Polycom new firmware and bootrom

Polycom are analy retentive about giving out software updates.

 -Original Message-
 From: Nathan Alberti [mailto:[EMAIL PROTECTED]
 Sent: Thursday, September 07, 2006 10:25 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Polycom new firmware and bootrom

 Stupid question where did you find it ?

 Looked at their site downloads and under the extranet site but could  
 only see old versions.

 Nathan.

 On 07/09/2006, at 10:21 AM, Chris Dos wrote:

  Well, it seems that Polycom has release new firmware 2.0.1 and  
  bootrom 3.2.2.
  I've proceded to upgrade all my ip430 phones because they were  
  essentially
  broken with the original firmware.

  All the phones boot up fine now, grab their files.  They 
 just won't  
  talk to the
  asterisk server any more.   I just figured out that I need to hard  
  code the sip
  server and tell it to talk udp only.  After this, the 
 phones worked  
  again.

  Any idea on what I need to configure to fix the phones so 
 they will  
  know which
  server to talk to and only talk to it via udp?

  Chris

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[asterisk-users] How to Install H323

2006-09-07 Thread Wasif

Hello,

Could anyone tell me how to install/configure H323 with Asterisk 1.2.11 .


Thanks

Wazb

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Re: [asterisk-users] using SIP to connect remote other VoIP server(Attn:Elpidio)

2006-09-07 Thread Crazy Boy

Hi Elpidio,I am Chandra from India. I have a doubt. I am trying to solve my problem from many days. But, I couldn't able to solve this problem. I am using Asterisk with Redhat Enterprise Linux 4.2. The problem is 5060 port is blocked. After stop my firewall (service iptables stop) also, 5060 port is not opening. I checked with the below command:# nmap -p5060 192.168.91.22---This is my IP addressand it is showing that port 5060 is closed. How can I enable and open this 5060 port? Really, I am breaking my head with this problem. SIP is not working because of this problem. Please tell me a solution. Looking forward to your reply. Thank you.Regards,Chandra.Elpidio Ramos [EMAIL PROTECTED] wrote: Hi,This is a sample file I am currently using on my
 server.  My server has a public IP address and an internal IP address (duan NIC).  It runs Fedora Core 3 running iptables firewall already configured with ports   4569, 5060, 1-2 open (udp and tcp)[general]context=defaultallowguest=norealm=your.hostname.extbindaddr=0.0.0.0bindport=5060externip=your.server.ip.addresssrvlookup=nomaxexpirey=3600disallow=allallow=ulawallow=ilbcallow=gsmmusicclass=defaultlanguage=esrtptimeout=120rtpholdtimeout=300useragent=asterisklocalnet=10.10.10.0/255.255.255.0rtcachefriends=noqualify=yes[311]type=friendregexten=311username=311secret=311callerid="User on extension 311" 311host=dynamicnat=yescanreinvite=no [312] type=friendregexten=312username=312secret=312callerid="User
 on extension 312" 312host=dynamicnat=yescanreinvite=no  tengulre [EMAIL PROTECTED] wrote:   How to using SIP to connect remote other VoIP server? is it only running one line voice if I registered a one SIP account?  anybody can give me some sample configuration files? thanks a lot!  ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options  visit:http://lists.digium.com/mailman/listinfo/asterisk-users   
   Elpidio Ramos PresidentRM International ServicesSA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax+52 (55)1755-6601 CellUSA: +1 (801) 494-1415 Office   +1 (240) 250-8264 Fax   +1 (801) 938-4740Direct  ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users 
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Re: [asterisk-users] Polycom new firmware and bootrom

Typically you have to go to a reseller who you purchased Polycom equipment from. Even then it can be tricky since they have to find away to get you the files with out upsetting Polycom.
On 9/7/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Polycom are analy retentive about giving out software updates. -Original Message- From: Nathan Alberti [mailto:[EMAIL PROTECTED]] Sent: Thursday, September 07, 2006 10:25 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom new firmware and bootrom Stupid question where did you find it ?
 Looked at their site downloads and under the extranet site but could only see old versions. Nathan. On 07/09/2006, at 10:21 AM, Chris Dos wrote:
  Well, it seems that Polycom has release new firmware 2.0.1 and  bootrom 3.2.2.  I've proceded to upgrade all my ip430 phones because they were  essentially  broken with the original firmware.
   All the phones boot up fine now, grab their files.They just won't  talk to the  asterisk server any more. I just figured out that I need to hard  code the sip
  server and tell it to talk udp only.After this, the phones worked  again.   Any idea on what I need to configure to fix the phones so they will  know which
  server to talk to and only talk to it via udp?   Chris   ___  --Bandwidth and Colocation provided by 
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Re: [asterisk-users] Polycom new firmware and bootrom

2006-09-07 Thread Jessee J Holmes

All authorized Polycom resellers will have access to this firmware and are required to provide this firmware to you. Contact the reseller you purchased the Polycom phone from. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 7, 2006, at 11:55 AM, Bruce Reeves wrote:Typically you have to go to a reseller who you purchased Polycom equipment from. Even then it can be tricky since they have to find away to get you the files with out upsetting Polycom. On 9/7/06, Douglas Garstang [EMAIL PROTECTED] wrote: Polycom are analy retentive about giving out software updates. -Original Message- From: Nathan Alberti [mailto:[EMAIL PROTECTED]] Sent: Thursday, September 07, 2006 10:25 AM  To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom new firmware and bootrom Stupid question where did you find it ?  Looked at their site downloads and under the extranet site but could only see old versions. Nathan. On 07/09/2006, at 10:21 AM, Chris Dos wrote:   Well, it seems that Polycom has release new firmware 2.0.1 and  bootrom 3.2.2.  I've proceded to upgrade all my ip430 phones because they were  essentially  broken with the original firmware.All the phones boot up fine now, grab their files.  They just won't  talk to the  asterisk server any more.   I just figured out that I need to hard  code the sip   server and tell it to talk udp only.  After this, the phones worked  again.   Any idea on what I need to configure to fix the phones so they will  know which   server to talk to and only talk to it via udp?   Chris   ___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com --  asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth 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 --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
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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 AMTo: 
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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RE: [asterisk-users] 0005162: RTP Packetization : Few questions

2006-09-07 Thread Dan Austin


 As far as the above is concerned I have the following:

 I am using Asterisk 1.2.10, patched with this patch for 1.2.10.
 I have 2 * boxes.  They call each other over SIP, and I have in 
 sip.conf on both boxes

 autoframing=yes
 disallow=all
 allow=g729:80

 When A calls B, it sets ptime:80.

 On B I see this:
 We're at 192.168.0.64 port 11004
 Adding codec 0x100 (g729) to SDP
 Sep  7 18:16:16 WARNING[5529]: frame.c:1072 ast_codec_pref_getsize:
 Framing not set for codec g729, using default 20 and ptime:20
I'll have a look at the 1.2.10 patch

 So B is setting packetization to 20, when it should be 80, and is not 
 respecting autoframing.
Another developer wrote the autoframing feature, and I have not used
it, but I'll look to see if there is an obvious reason why it does
not find or honor the ptime.

Can you capture the SIP INVITE dialog on box B so I can see the SDP
offer, and look to see if the ptime element is present and set
properly?

 I have tried this with reinvites=yes and no, and autoframing=yes and 
 no, still the same.
Can you try with autoframing=no and force 80ms on both sides?

Also, I am not sure if this is a bug.
If in sip.conf, if I set
 
 
[yusuf]
username=yusuf
secret=yusuf
type=friend
callerid=1002
nat=yes
canreinvite=no
allow=all
host=dynamic
context=sip
 
 
 BUG!
 Which version of the patch and what SVN version?  I suspect it has
 to do with one or more of the codecs that we could not find
 framing/packetization details about.  Is alaw the codec used in the
 call that causes the crash?
 
 
then when asterisk calls, it says I have not set Framing (like above
 
 msg),
 
then asterisk just dies.
 
 
If I chane the line
allow=all to allow=alaw:20
 
 
then its fine, and asterisk does not die.
 
 
Dont know if this is a bug, so I wont post debug/full messages now.
 



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RE: [asterisk-users] Re: Asterisk hangs up after 10-15 minutes when SIPPhone is on mute

2006-09-07 Thread Mike

Thanks Tony.  Its possible that the phone stops sending RTP stream (but it
certainly is receiving some!). How do I get Asterisk to stop caring whether
it receives RTP or not?

Yes there is a NAT between the phone the the Internet.  The Asterisk server
doesn't have NAT though.

I'll try to find out what I can from my limited RTP expertise. I appreciate
the response and the hints.

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: September 7, 2006 12:09 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Asterisk hangs up after 10-15 minutes when
SIPPhone is on mute

In article [EMAIL PROTECTED],
Mike [EMAIL PROTECTED] wrote:
  
 I have a Polycom 501 connected to Asterisk 1.2.4 (and then connected 
 to a VOIP provider, Unlimitel in my case).  My job requires me to 
 attend conference calls regularly, and I am usually there as a silent
listener.
 Therefore, I mute my phone.
  
 I`ve noticed that if I mute my phone, after 10-15 of being muted, the 
 line hangs up.  I had the same problem with my GXP-2000 before, so I 
 dismissed the phone as being the problem.  If I unmute regularly (or 
 the entire time), the line doesnt hang up (until it reaches max 
 timeout of course, which is much more than 15 minutes).  So the 
 problem is my phone is muted. I have observed that about 6 times (out 
 of 6 tries) in the last 4 months.  It`s a reccuring issue for sure.

Can you user Ethereal or similar to find out whether the phone stops sending
the RTP stream to Asterisk when it is muted?

Is there any kind of NAT gateway between the phone and Asterisk?

If both of the above are true, it might be possible that the NAT gateway's
session mapping for that stream is timing out.

Or perhaps Asterisk decides to hang up the call if it doesn't get any RTP
for a certain length of time.

Just a couple of ideas to try.

If the phone is still sending RTP when muted, the stream presumably contains
perfect silence. I don't know if that gets detected anywhere.

Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] Asterisk and NAT ?

2006-09-07 Thread Noc Phibee


yusuf a écrit :


Hi,

you dont have to/should'nt be using different SIP ports for each 
phone.  Its completely not needed.  Also, you dont have/need to port 
forward.  Just open ports 5060 and 1000-2, on the box that 
asterisk is running, and on your NAT router. Dont port forward.


Then in sip.conf


 [202]
 username=202
 secret=X
 type=friend
 host=dynamic
 disallow=all
 allow=g729
 allow=alaw
 allow=ulaw
 context=interne
 nat=yes
 canreinvite=no   



 [200]
 username=200
 secret=X
 type=friend
 host=dynamic
 disallow=all
 allow=g729
 allow=alaw
 allow=ulaw
 context=interne
 nat=yes
 canreinvite=no   



then restart linksys and thomson, and you will see that they both 
register on asterisk cli.  Now you will be able to call/receive on both.






Thanks for your answer, but if i don't put a port forward, i have :

200/20083.167.122.119   D   N  5060 UNREACHABLE

On the thomson, i have SIP Unregister, it's a important option  ?

Thanks bye


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RE: [asterisk-users] Polycom new firmware and bootrom




That 
process is worse than pulling teeth!

  -Original Message-From: Jessee J Holmes 
  [mailto:[EMAIL PROTECTED]Sent: Thursday, September 07, 2006 
  11:25 AMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [asterisk-users] Polycom new firmware and 
  bootromAll authorized Polycom resellers will have access 
  to this firmware and are required to provide this firmware to you. Contact the 
  reseller you purchased the Polycom phone from.
  
  
  
  Jessee Holmes
  Atacomm / Ataractic Corporation
  www.atacomm.com
  V: 1-877-700-VOIP
  [EMAIL PROTECTED]
  
  Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/
  
  On Sep 7, 2006, at 11:55 AM, Bruce Reeves wrote:
  Typically you have to go to a reseller who you 
purchased Polycom equipment from. Even then it can be tricky since they have 
to find away to get you the files with out upsetting Polycom.
On 9/7/06, Douglas 
Garstang [EMAIL PROTECTED] 
wrote:
Polycom 
  are analy retentive about giving out software updates. 
  -Original Message- From: Nathan Alberti [mailto:[EMAIL PROTECTED]] Sent: 
  Thursday, September 07, 2006 10:25 AM  To: Asterisk Users Mailing 
  List - Non-Commercial Discussion Subject: Re: [asterisk-users] 
  Polycom new firmware and bootrom 
  Stupid question where did you find it ?  
  Looked at their site downloads and under the extranet site but 
  could only see old versions. 
  Nathan. On 07/09/2006, at 10:21 AM, Chris Dos 
  wrote:  Well, it seems that Polycom has release new 
  firmware 2.0.1 and  bootrom 3.2.2.  I've proceded 
  to upgrade all my ip430 phones because they were  
  essentially  broken with the original firmware.  
All the phones boot up fine now, grab their 
  files.They just won't  talk to the 
   asterisk server any more. I just figured out that I need 
  to hard  code the sip   server and tell it to talk 
  udp only.After this, the phones worked  
  again.   Any idea on what I need to configure to 
  fix the phones so they will  know which   
  server to talk to and only talk to it via udp?  
   Chris   
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Re: [asterisk-users] Polycom new firmware and bootrom

2006-09-07 Thread Brandon Galbraith

You've never tried to get firmware for the Cisco 7960 I take it? =) I'd rather try to write it myself then go through that again.-brandonOn 9/7/06, 
Douglas Garstang [EMAIL PROTECTED] wrote:







That 
process is worse than pulling teeth!

  -Original Message-From: Jessee J Holmes 
  [mailto:[EMAIL PROTECTED]]Sent: Thursday, September 07, 2006 
  11:25 AMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [asterisk-users] Polycom new firmware and 
  bootromAll authorized Polycom resellers will have access 
  to this firmware and are required to provide this firmware to you. Contact the 
  reseller you purchased the Polycom phone from.
  
  
  

  
Jessee Holmes
  
Atacomm / Ataractic Corporation
  
www.atacomm.com
  
V: 1-877-700-VOIP
  
[EMAIL PROTECTED]
  

  
Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/

  
  On Sep 7, 2006, at 11:55 AM, Bruce Reeves wrote:
  Typically you have to go to a reseller who you 
purchased Polycom equipment from. Even then it can be tricky since they have 
to find away to get you the files with out upsetting Polycom.
On 9/7/06, Douglas 
Garstang [EMAIL PROTECTED] 
wrote:
Polycom 
  are analy retentive about giving out software updates. 
  -Original Message- From: Nathan Alberti [mailto:[EMAIL PROTECTED]] Sent: 
  Thursday, September 07, 2006 10:25 AM  To: Asterisk Users Mailing 
  List - Non-Commercial Discussion Subject: Re: [asterisk-users] 
  Polycom new firmware and bootrom 
  Stupid question where did you find it ?  
  Looked at their site downloads and under the extranet site but 
  could only see old versions. 
  Nathan. On 07/09/2006, at 10:21 AM, Chris Dos 
  wrote:  Well, it seems that Polycom has release new 
  firmware 2.0.1 and  bootrom 3.2.2.  I've proceded 
  to upgrade all my ip430 phones because they were  
  essentially  broken with the original firmware.  
All the phones boot up fine now, grab their 
  files.They just won't  talk to the 
   asterisk server any more. I just figured out that I need 
  to hard  code the sip   server and tell it to talk 
  udp only.After this, the phones worked  
  again.   Any idea on what I need to configure to 
  fix the phones so they will  know which   
  server to talk to and only talk to it via udp?  
   Chris   
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Re: [asterisk-users] How to Install H323

2006-09-07 Thread Alberto Sagredo

I think remember there is a readme on /docs that talks about 
chan_h323.Check it !


Anyway you could try too at voip.info dot org.

Regards


Wasif escribió:

Hello,

Could anyone tell me how to install/configure H323 with Asterisk 1.2.11 .


Thanks

Wazb

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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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Re: [asterisk-users] using SIP to connect remote other VoIP server(Attn:Elpidio)

2006-09-07 Thread Rich Adamson


Crazy Boy wrote:

Hi Elpidio,

I am Chandra from India. I have a doubt. I am trying to solve my problem 
from many days. But, I couldn't able to solve this problem. I am using 
Asterisk with Redhat Enterprise Linux 4.2. The problem is 5060 port is 
blocked. After stop my firewall (service iptables stop) also, 5060 port 
is not opening. I checked with the below command:

# nmap -p5060 192.168.91.22---This is my IP address
and it is showing that port 5060 is closed. How can I enable and open 
this 5060 port? Really, I am breaking my head with this problem. SIP is 
not working because of this problem. Please tell me a solution. Looking 
forward to your reply. Thank you.


The quickest way to determine whether an application is listening on a 
port is to simply do a 'netstat -an' from the linux command line. You 
should see something like this:

 udp0  0 0.0.0.0:50600.0.0.0:*

If you don't see that, then asterisk is not opening the port.

From an asterisk command line, do 'show modules like sip' and you 
should see something like this:
 Module Description 
  Use Count

 chan_sip.soSession Initiation Protocol (SIP)0

If you don't see that, then asterisk is not loading the chan_sip.so 
module for some reason.


Look in /etc/asterisk/modules.conf and make sure there is NOT an entry 
in that file that looks something like this:

 noload = chan_sip.so

If that entry is not there, then you either have a problem with the 
configuration of the file /etc/asterisk/sip.conf, or, some other problem 
that is causing asterisk to not load chan_sip.so.


If you are sure the sip.conf is absolutely correct and error free, then 
stop asterisk, and start it from the linux command line with 'asterisk 
-c'. There should be some indication why chan_sip.so is not be loaded, etc.


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




Hello Michael,

I just had both Mom and my brother up as extensions on my Asterisk pbx 
using IAX2, the Cubix phone for now, but I downloaded and tried several. I 
loke 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 rest 
off.


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 Ellson
CCDA, 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 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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   http://lists.digium.com/mailman/listinfo/asterisk-users


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--
Bruce

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 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

Re: [asterisk-users] Polycom new firmware and bootrom

2006-09-07 Thread Jessee J Holmes

I agree, Polycom should make this publicly available; but unfortunately, I've seen worse policies out there *cough* Cisco *cough*.The reseller shouldn't give you any hassle about it  and if they do, or if you can't reach them for whatever reason (A.K.A. no email replies or phones being answered), that's violation of the contract they had to sign to be Polycom Authorized in the first place, and Polycom will take immediate action to rectify the situation if that's the case. I'd suggest finding another place to purchase from if your current reseller is giving you troubles with getting the firmware for you.The firmware is probably about a 50 MB download (i think) and can be downloaded via HTTP or FTP. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 7, 2006, at 1:19 PM, Douglas Garstang wrote: That process is worse than pulling teeth!   -Original Message-From: Jessee J Holmes   [mailto:[EMAIL PROTECTED]]Sent: Thursday, September 07, 2006   11:25 AMTo: Asterisk Users Mailing List - Non-Commercial   DiscussionSubject: Re: [asterisk-users] Polycom new firmware and   bootromAll authorized Polycom resellers will have access   to this firmware and are required to provide this firmware to you. Contact the   reseller you purchased the Polycom phone from.  Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/On Sep 7, 2006, at 11:55 AM, Bruce Reeves wrote:  Typically you have to go to a reseller who you purchased Polycom equipment from. Even then it can be tricky since they have to find away to get you the files with out upsetting Polycom.On 9/7/06, Douglas Garstang [EMAIL PROTECTED] wrote:Polycom   are analy retentive about giving out software updates.   -Original Message- From: Nathan Alberti [mailto:[EMAIL PROTECTED]] Sent:   Thursday, September 07, 2006 10:25 AM  To: Asterisk Users Mailing   List - Non-Commercial Discussion Subject: Re: [asterisk-users]   Polycom new firmware and bootrom   Stupid question where did you find it ?Looked at their site downloads and under the extranet site but   could only see old versions.   Nathan. On 07/09/2006, at 10:21 AM, Chris Dos   wrote:  Well, it seems that Polycom has release new   firmware 2.0.1 and  bootrom 3.2.2.  I've proceded   to upgrade all my ip430 phones because they wereessentially  broken with the original firmware.  All the phones boot up fine now, grab their   files.  They just won't  talk to theasterisk server any more.   I just figured out that I need   to hard  code the sip   server and tell it to talk   udp only.  After this, the phones workedagain.   Any idea on what I need to configure to   fix the phones so they will  know which server to talk to and only talk to it via udp? Chris ___  --Bandwidth   and Colocation provided by Easynews.com   --   asterisk-users mailing list  To   UNSUBSCRIBE or update options visit:      http://lists.digium.com/mailman/listinfo/asterisk-users   ___ --Bandwidth and   Colocation provided by Easynews.com   -- asterisk-users mailing list To UNSUBSCRIBE or   update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users   ___--Bandwidth   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 --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
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RE : RE: [asterisk-users] Polycom new firmware and bootrom

2006-09-07 Thread harrygaillac-sip

I have access to the ftp server of polycom
--- Douglas Garstang [EMAIL PROTECTED] a écrit
:

 That process is worse than pulling teeth!

 -Original Message-
 From: Jessee J Holmes [mailto:[EMAIL PROTECTED]
 Sent: Thursday, September 07, 2006 11:25 AM
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [asterisk-users] Polycom new firmware
 and bootrom

 All authorized Polycom resellers will have access to
 this firmware and are required to provide this
 firmware to you. Contact the reseller you purchased
 the Polycom phone from. 

 Jessee Holmes

 Atacomm / Ataractic Corporation

 www.atacomm.com

 V: 1-877-700-VOIP

 [EMAIL PROTECTED]

 Looking for voice over IP products?  Visit our VoIP
 store at http://voipstore.atacomm.com/

 On Sep 7, 2006, at 11:55 AM, Bruce Reeves wrote:

 Typically you have to go to a reseller who you
 purchased Polycom equipment from. Even then it can
 be tricky since they have to find away to get you
 the files with out upsetting Polycom.

 On 9/7/06, Douglas Garstang 
 [EMAIL PROTECTED] wrote: 

 Polycom are analy retentive about giving out
 software updates.

  -Original Message-
  From: Nathan Alberti [mailto:
 [EMAIL PROTECTED]
  Sent: Thursday, September 07, 2006 10:25 AM 
  To: Asterisk Users Mailing List - Non-Commercial
 Discussion
  Subject: Re: [asterisk-users] Polycom new firmware
 and bootrom

  Stupid question where did you find it ? 

  Looked at their site downloads and under the
 extranet site but could
  only see old versions.

  Nathan.

  On 07/09/2006, at 10:21 AM, Chris Dos wrote:

   Well, it seems that Polycom has release new
 firmware 2.0.1 and
   bootrom 3.2.2.
   I've proceded to upgrade all my ip430 phones
 because they were
   essentially
   broken with the original firmware. 

   All the phones boot up fine now, grab their
 files.  They
  just won't
   talk to the
   asterisk server any more.   I just figured out
 that I need to hard
   code the sip 
   server and tell it to talk udp only.  After
 this, the
  phones worked
   again.

   Any idea on what I need to configure to fix the
 phones so
  they will
   know which 
   server to talk to and only talk to it via udp?

   Chris

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 Nortex Networks 
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Re: [asterisk-users] Re: Volume events causing talk off on Asterisk with Digium 411P

2006-09-07 Thread Zoa



But does it help ? Is it better than before ?
Do you have a good way of debugging ? (like an audio recording that i 
could play ?)

Does it show something on the cli when it happens ?

Zoa

Servetas, Andrew wrote:

They recommended changing the default value of 1000 up or down 
incrementally until it works better. We’re currently at 2000, and 
we’re still not completely free of events.




What were the proposed changes to VPM_DEFAULT_DTMFTHRESHOLD ?



--

--

Steven



http://www.glimasoutheast.org







 Servetas, Andrew andrew.servetas at dirigosoft.com 
http://lists.digium.com/mailman/listinfo/asterisk-users wrote in message 
news:28289145AD231E418DB8CABE0BE392AA016F8176 at casco.stroudwater.net... 
http://lists.digium.com/mailman/listinfo/asterisk-users

  




 We are experiencing random talk off events when we hear a loud volume event on 
the PSTN side of our calls.  We do not always hear the spurious DTMF, but I can 
see it in the console when I have the debug and verbose levels turned up.  We 
do however always have the associated brief periods of silence that immediately 
follow.  Sometimes they are only a matter of seconds, other times they can be 
as long as a minute.  We hear it most often if the remote party is on a 
cellular phone with a lot of background noise, or if a loud noise happens 
during the call.  Neither party can hear the other when this happens.  It 
almost reacts like an AGC circuit is muting the call.



  




 We are using a Digium TE411P quad-span T1 card on 1.2.5.  I called Digium 
support and we have played with the VPM_DEFAULT_DTMFTHRESHOLD in the WCT4XXP.C 
driver as recommended, and all the RELAXDTMF and GAIN settings in Zapata.conf 
are set according to their recommendations.



  




 Has anyone else experienced this, and if so, what have you done to correct it?



  




 Andy Servetas



 CTI Support Engineer



   




 Dirigosoft Corporation



 Portland, ME



  




 www.dirigosoft.com  




  








--





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Re: [asterisk-users] Polycom new firmware and bootrom

2006-09-07 Thread Jessee J Holmes

Also keep in mind that as of right now, the latest bootrom and firmware available from Polycom (and thus your reseller) are Bootrom 3.2.2 and Firmware 2.0.1The 2.0.1 firmware is new as of a day or two and include some enhancements for buddy lists and shared presence as well as newly added secured TLS support (according to Polycom). Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 7, 2006, at 1:19 PM, Douglas Garstang wrote: That process is worse than pulling teeth!   -Original Message-From: Jessee J Holmes   [mailto:[EMAIL PROTECTED]]Sent: Thursday, September 07, 2006   11:25 AMTo: Asterisk Users Mailing List - Non-Commercial   DiscussionSubject: Re: [asterisk-users] Polycom new firmware and   bootromAll authorized Polycom resellers will have access   to this firmware and are required to provide this firmware to you. Contact the   reseller you purchased the Polycom phone from.  Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/On Sep 7, 2006, at 11:55 AM, Bruce Reeves wrote:  Typically you have to go to a reseller who you purchased Polycom equipment from. Even then it can be tricky since they have to find away to get you the files with out upsetting Polycom.On 9/7/06, Douglas Garstang [EMAIL PROTECTED] wrote:Polycom   are analy retentive about giving out software updates.   -Original Message- From: Nathan Alberti [mailto:[EMAIL PROTECTED]] Sent:   Thursday, September 07, 2006 10:25 AM  To: Asterisk Users Mailing   List - Non-Commercial Discussion Subject: Re: [asterisk-users]   Polycom new firmware and bootrom   Stupid question where did you find it ?Looked at their site downloads and under the extranet site but   could only see old versions.   Nathan. On 07/09/2006, at 10:21 AM, Chris Dos   wrote:  Well, it seems that Polycom has release new   firmware 2.0.1 and  bootrom 3.2.2.  I've proceded   to upgrade all my ip430 phones because they wereessentially  broken with the original firmware.  All the phones boot up fine now, grab their   files.  They just won't  talk to theasterisk server any more.   I just figured out that I need   to hard  code the sip   server and tell it to talk   udp only.  After this, the phones workedagain.   Any idea on what I need to configure to   fix the phones so they will  know which server to talk to and only talk to it via udp? Chris ___  --Bandwidth   and Colocation provided by Easynews.com   --   asterisk-users mailing list  To   UNSUBSCRIBE or update options visit:      http://lists.digium.com/mailman/listinfo/asterisk-users   ___ --Bandwidth and   Colocation provided by Easynews.com   -- asterisk-users mailing list To UNSUBSCRIBE or   update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users   ___--Bandwidth   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 --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
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[asterisk-users] Re: bristuff compile problems with kernel 2.6.17.11

2006-09-07 Thread Arik Raffael Funke


Tzafrir Cohen wrote:

On Thu, Sep 07, 2006 at 01:27:36PM +0200, Arik Raffael Funke wrote:

Hi,

has anybody had success compiling bristuff with kernel 2.6.17.11? Error 
messages are below...


This is not code that is touched by the bristuff patch.

Anyway, I'd try the latest 0.3.0 bristuff patch.


Where is the code from then? I thought zaptel was part of bristuff. 
Anyway, has anybody been able to get it to work with kernel-2.6.17.11?


I am not using bristuff 0.3.0 because it requires asterisk 1.2. All my 
configuration currently bases on 1.0 and I am not quite ready to move it 
to 1.2.


Cheers,
Arik

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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 and

Re: [asterisk-users] using SIP to connect remote other VoIP server(Attn:Elpidio)

2006-09-07 Thread Elpidio Ramos

I just went thru the same problem days ago and it all ended being problems with the firewall.Even if the application is listening in a given port, that doesn't mean the port is open in the firewall.try thise to see if the firewall is letting the traffic thru an specific port:iptables -LYou should see something like: ACCEPT udp -- anywhere anywhere state NEW udp dpt:sip orACCEPT udp -- anywhere anywhere state NEW udp
dpt:5060The same applies to the iax2 port or the rtp port. You can use the iptables to open new ports or use the graphical tool in the gnome graphical environment to configure the firewall.Make sure you reboot the machine after your changes are made. It is the best way to ensure the new configuration takes place./etc/init.d/iptables stop /etc/init.d/iptables start those commans help you stop or start the firewall.ElpidioRich Adamson [EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi Elpidio, I am Chandra from India. I have a doubt. I am trying to solve my problem from many days. But, I couldn't able to solve this problem. I am
using Asterisk with Redhat Enterprise Linux 4.2. The problem is 5060 port is blocked. After stop my firewall (service iptables stop) also, 5060 port is not opening. I checked with the below command: # nmap -p5060 192.168.91.22---This is my IP address and it is showing that port 5060 is closed. How can I enable and open this 5060 port? Really, I am breaking my head with this problem. SIP is not working because of this problem. Please tell me a solution. Looking forward to your reply. Thank you.The quickest way to determine whether an application is listening on a port is to simply do a 'netstat -an' from the linux command line. You should see something like this:udp 0 0 0.0.0.0:5060 0.0.0.0:*If you don't see that, then asterisk is not opening the port.From an asterisk command line, do 'show modules like sip' and you should see something like this:Module
Description Use Countchan_sip.so Session Initiation Protocol (SIP) 0If you don't see that, then asterisk is not loading the chan_sip.so module for some reason.Look in /etc/asterisk/modules.conf and make sure there is NOT an entry in that file that looks something like this:noload = chan_sip.soIf that entry is not there, then you either have a problem with the configuration of the file /etc/asterisk/sip.conf, or, some other problem that is causing asterisk to not load chan_sip.so.If you are sure the sip.conf is absolutely correct and error free, then stop asterisk, and start it from the linux command line with 'asterisk -c'. There should be some indication why chan_sip.so is not be loaded, etc.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options
visit:http://lists.digium.com/mailman/listinfo/asterisk-users Elpidio Ramos PresidentRM International ServicesSA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax+52 (55)1755-6601 CellUSA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740Direct ___
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RE: [asterisk-users] 0005162: RTP Packetization : Few questions

2006-09-07 Thread Dan Austin

2006-08-31 22:11:22 WARNING[1278]: frame.c:1072 
ast_codec_pref_getsize: Framing not set for codec alaw, using 
default 20
 
 As far as the above is concerned I have the following:

 I am using Asterisk 1.2.10, patched with this patch for 1.2.10.  
 I have 2 * boxes.  They call each other over SIP, and I have in 
 sip.conf on both boxes

 autoframing=yes
 disallow=all
 allow=g729:80

 When A calls B, it sets ptime:80.

 On B I see this:
 We're at 192.168.0.64 port 11004
 Adding codec 0x100 (g729) to SDP
 Sep  7 18:16:16 WARNING[5529]: frame.c:1072 ast_codec_pref_getsize: 
 Framing not set for codec g729, using default 20 and ptime:20

 So B is setting packetization to 20, when it should be 80, and is 
 not respecting autoframing.

 I have tried this with reinvites=yes and no, and autoframing=yes and
 no, still the same.

The autoframing patch forgot to remove an earlier check for 'ptime'
in the SDP that would cause chan_sip to ignore the ptime value.

I am working on trunk, so the line numbers may not match up, but
near line 4748 you will should find this block of code:

} else if (!strncasecmp(a, ptime:, (size_t) 6)) {
if (debug)
ast_verbose(Got unsupported
a:ptime in SDP offer \n);
breakout = TRUE;

Simply comment out the breakout = TRUE; line like this.

} else if (!strncasecmp(a, ptime:, (size_t) 6)) {
if (debug)
ast_verbose(Got unsupported
a:ptime in SDP offer \n);
/* breakout = TRUE; */

That fixes up autoframing in my tests, if it works for you, I will
prepare a proper patch.

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

The configuration is done in the softphone, like Nick mentions then you can tweak it with a text editor per individual.On 9/7/06, Ferguson, Michael 
[EMAIL PROTECTED] 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 
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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-- BruceNortex Networks 
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Re: [asterisk-users] Polycom new firmware and bootrom

2006-09-07 Thread Chris Dos

Not to mention the feature that the new firmware and bootrom that prevent it
from registering with the Asterisk server unless you hard code the sip settings.

Chris

Jessee J Holmes wrote:
 Also keep in mind that as of right now, the latest bootrom and firmware
 available from Polycom (and thus your reseller) are Bootrom 3.2.2 and
 Firmware 2.0.1
 
 The 2.0.1 firmware is new as of a day or two and include some
 enhancements for buddy lists and shared presence as well as newly added
 secured TLS support (according to Polycom).
 
 
 
 Jessee Holmes
 
 Atacomm / Ataractic Corporation
 
 www.atacomm.com
 
 V: 1-877-700-VOIP
 
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 
 
 Looking for voice over IP products?  Visit our VoIP store at
 http://voipstore.atacomm.com/
 
 
 
 On Sep 7, 2006, at 1:19 PM, Douglas Garstang wrote:
 
 That process is worse than pulling teeth!

 -Original Message-
 *From:* Jessee J Holmes [mailto:[EMAIL PROTECTED]
 *Sent:* Thursday, September 07, 2006 11:25 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Polycom new firmware and bootrom

 All authorized Polycom resellers will have access to this firmware
 and are required to provide this firmware to you. Contact the
 reseller you purchased the Polycom phone from.


 Jessee Holmes
 Atacomm / Ataractic Corporation
 www.atacomm.com
 V: 1-877-700-VOIP
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]

 Looking for voice over IP products?  Visit our VoIP store at
 http://voipstore.atacomm.com/


 On Sep 7, 2006, at 11:55 AM, Bruce Reeves wrote:

 Typically you have to go to a reseller who you purchased Polycom
 equipment from. Even then it can be tricky since they have to
 find away to get you the files with out upsetting Polycom.

 On 9/7/06, *Douglas Garstang* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:

 Polycom are analy retentive about giving out software updates.

  -Original Message-
  From: Nathan Alberti [mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]]
  Sent: Thursday, September 07, 2006 10:25 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Polycom new firmware and bootrom
 
 
 
 
  Stupid question where did you find it ?
 
 
  Looked at their site downloads and under the extranet site
 but could
  only see old versions.
 
  Nathan.
 
  On 07/09/2006, at 10:21 AM, Chris Dos wrote:
 
   Well, it seems that Polycom has release new firmware
 2.0.1 and
   bootrom 3.2.2.
   I've proceded to upgrade all my ip430 phones because they
 were
   essentially
   broken with the original firmware.
  
   All the phones boot up fine now, grab their files.  They
  just won't
   talk to the
   asterisk server any more.   I just figured out that I
 need to hard
   code the sip
   server and tell it to talk udp only.  After this, the
  phones worked
   again.
  
   Any idea on what I need to configure to fix the phones so
  they will
   know which
   server to talk to and only talk to it via udp?
  
   Chris
  
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 Nortex Networks
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Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.

2006-09-07 Thread Blake Krone

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 watchmy
ACL 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 video working as
well, h.263 I believe it is).Nick ___
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[asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?

2006-09-07 Thread Rushowr

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hey all,

I'm looking into setting up a system or two with either IMAP or ODBC
storage of Voicemail messages and wanted to hear about your experiences,
gather tips or warnings, etc, before I go diving too deep into it. Are
either of those storage methods working reliably for any of you? What
are some of the issues you had to deal with when setting it up? What's
the performance like? You get the general idea...

Quick stats on base test systems:

Latest SVN trunk as of this morning
Gentoo
MySQL 5.0
Realtime sip and iax peers/users
Realtime sip/iax/voicemail config
LARGE dialplan

Thanks in advance for any input,
SKM

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RE: [asterisk-users] Polycom new firmware and bootrom




The 
reseller doesn't hassle us... it just takes them several days to fulfill simple 
requests.


  -Original Message-From: Jessee J Holmes 
  [mailto:[EMAIL PROTECTED]Sent: Thursday, September 07, 2006 
  12:45 PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [asterisk-users] Polycom new firmware and 
  bootromI agree, Polycom should make this publicly 
  available; but unfortunately, I've seen worse policies out there *cough* Cisco 
  *cough*.
  
  The reseller shouldn't give you any hassle about it  and if they do, 
  or if you can't reach them for whatever reason (A.K.A. no email replies or 
  phones being answered), that's violation of the contract they had to sign to 
  be Polycom Authorized in the first place, and Polycom will take immediate 
  action to rectify the situation if that's the case. I'd suggest finding 
  another place to purchase from if your current reseller is giving you troubles 
  with getting the firmware for you.
  
  The firmware is probably about a 50 MB download (i think) and can be 
  downloaded via HTTP or FTP.
  
  
  
  Jessee Holmes
  Atacomm / Ataractic Corporation
  www.atacomm.com
  V: 1-877-700-VOIP
  [EMAIL PROTECTED]
  
  Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/
  
  On Sep 7, 2006, at 1:19 PM, Douglas Garstang wrote:
  
That process is worse than pulling teeth!

  -Original Message-From: Jessee J Holmes [mailto:[EMAIL PROTECTED]]Sent: 
  Thursday, September 07, 2006 11:25 AMTo: Asterisk Users Mailing 
  List - Non-Commercial DiscussionSubject: Re: [asterisk-users] 
  Polycom new firmware and bootromAll authorized 
  Polycom resellers will have access to this firmware and are required to 
  provide this firmware to you. Contact the reseller you purchased the 
  Polycom phone from. 
  
  
  
  Jessee Holmes
  Atacomm / Ataractic Corporation
  www.atacomm.com
  V: 1-877-700-VOIP
  [EMAIL PROTECTED]
  
  Looking for voice over IP products? Visit our VoIP store at 
  http://voipstore.atacomm.com/
  
  On Sep 7, 2006, at 11:55 AM, Bruce Reeves wrote:
  Typically you have to go to a reseller who you 
purchased Polycom equipment from. Even then it can be tricky since they 
have to find away to get you the files with out upsetting 
Polycom.
On 9/7/06, Douglas Garstang [EMAIL PROTECTED] 
wrote: 
Polycom 
  are analy retentive about giving out software updates. 
  -Original Message- From: Nathan Alberti [mailto:[EMAIL PROTECTED]] 
  Sent: Thursday, September 07, 2006 10:25 AM  To: Asterisk 
  Users Mailing List - Non-Commercial Discussion Subject: Re: 
  [asterisk-users] Polycom new firmware and 
  bootrom Stupid question 
  where did you find it ?  Looked at their site 
  downloads and under the extranet site but could only see old 
  versions. Nathan. On 07/09/2006, at 
  10:21 AM, Chris Dos wrote:  Well, it seems that 
  Polycom has release new firmware 2.0.1 and  bootrom 
  3.2.2.  I've proceded to upgrade all my ip430 phones 
  because they were  essentially  broken with 
  the original firmware.All the phones boot 
  up fine now, grab their files.They just 
  won't  talk to the  asterisk server any 
  more. I just figured out that I need to hard  
  code the sip   server and tell it to talk udp 
  only.After this, the phones worked  
  again.   Any idea on what I need to configure 
  to fix the phones so they will  know which 
server to talk to and only talk to it via udp? 
Chris  
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Re: [asterisk-users] Capacity for transcode G711 to G729

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

RR wrote:
 Hi matt,
 
 sorry this might be a stupid question but is a bit pertinent to me,
 I'd asked something similar in one of my last email regarding SMP. Do
 you know if (*) is capable of making use of HT support i.e is
 multi-threaded and improves performance for operations like
 transcoding? Is that a valid question or is this only dependant on the

I don't think you will get double or anything, in fact many people have
suggested that HT be turned off when people experience problems.

 OS/Kernel, the CPU itself and the chipset on the motherboard? If I
 boot into an SMP kernel with Asterisk compiled with the SMP kernel
 source, would it just make use of multi-threading as the load
 increases on cpu-intensive operations?

The best use I have seen is the newly converted IAX2 which can use
multithreading in version 1.4, the beta of which should be released
later this week.

The best idea would be to compile Asterisk, run some tests (show
translation recalc 60) with HT turned on, restart the box, bring it up
with HT turned off and try again.

You should also run a few calls and check the CPU.

 Also, when you said the normal is 120 simultaneous transcoding
 operations, what is normal? I have a P4 w/HT 3.4Ghz, 2GB RAM
 machine. Would that be above or below normal?
 
 Thanks much
 \R

I would think that is above normal but not by much, I'm not sure what
normal was, nor can I find the Digium document where this was stated.

It wasn't that long ago.

I'm doing some more tests on a 3000 line setup (external DS3s via
Asterisk and SER clusters) at the moment which we are splitting to be
half G.729 and half ulaw, and I will try to post some results.

- --
Cheers,

Matt Riddell
___

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http://wap.sineapps.com (Daily Asterisk News for your cellphone)
http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
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Re: [asterisk-users] Asterisk Outgoing Spool Failed

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Arun Kumar wrote:
 hi
 
 my asterisk -r shows me Most of the times Outgoing Spool Failed. Can some
 one tell me why is it happening and how to solve this issue. Is it a
 problem
 ?

You'd need to provide more information.

Does it work when you call normally?

Are you spooling lots of calls at the same time?

Show us your spool file.

- --
Cheers,

Matt Riddell
___

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http://wap.sineapps.com (Daily Asterisk News for your cellphone)
http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
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Re: [asterisk-users] svn trunk or branches ???

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Ronald Wiplinger wrote:
 My last update was a while back and as I remember svn trunk did not
 compile and I was advised to use branches 1.2 till further notice.
 
 Have I missed the further notice and can we use now svn trunk or is the
 advice still to use branches 1.2 ???

Use svn branch 1.2 for now.

A beta of 1.4 (svn trunk) will be released later this week.

If you're happy to be using a beta, use it when released later this week.

After 1.4 is released, 1.4 will be the STABLE branch (i.e. only
bugfixes) and SVN trunk will be the code that will be released as 1.6.

- --
Cheers,

Matt Riddell
___

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http://wap.sineapps.com (Daily Asterisk News for your cellphone)
http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
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Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.



HUSHshout I think it was called...

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


On Thu, 7 Sep 2006, Blake Krone wrote:


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 pbx
 using IAX2, the Cubix phone for now, but I downloaded and tried several. I
 loke 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 rest
 off.

 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 Ellson
 CCDA, 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 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

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