[asterisk-users] loosing Sipura 841 almost exactly on the hour

[Jim Sturtevant]

I've got a curious situation.  I have a client that is adding a new 
site.  At the moment they have a single 841 at the remote locaiton.  The 
Asterisk PBX is on a public IP simple iptables firewall.


On the remote location where the SPA841 is located they're running a 
Win2003 ISA server as a router/firewall (ugh!).  I would think this is a 
firewall setting, but i can't imagine what it is.  I would also have 
thought the asterisk qualify pings would have kept the connection live.


I have asterisk qualify on, thus the messages below.  


Thanks in advance,
Jim




Sep 12 15:33:20 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 48
Sep 12 16:34:40 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 57
Sep 12 17:40:55 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 64
Sep 12 18:41:45 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 302
Sep 12 19:42:53 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 114
Sep 12 20:44:03 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 106
Sep 12 21:45:14 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 96
Sep 12 22:46:10 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 102
Sep 12 23:47:25 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 88
Sep 13 00:48:26 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 115
Sep 13 01:49:29 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 88
Sep 13 02:50:32 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 68
Sep 13 03:51:36 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 119
Sep 13 04:52:52 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 115
Sep 13 05:53:58 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 101
Sep 13 06:55:05 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 86
Sep 13 07:56:19 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 105
Sep 13 08:57:24 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 179
Sep 13 09:58:26 NOTICE[13331] chan_sip.c: Peer '220' is now 
UNREACHABLE!  Last qualify: 136

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[Asterisk-Users] *81, block CID, using ATA

[Jim Sturtevant]

I searched the wiki for a solution to allow a user on an
analog ATA to send *81 to block Asterisk CID (or any other * code).

 

The ATA has *81 built in to block the CID the ATA generates.

 

Any examples would be appreciated.

 






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[Asterisk-Users] *81, block CID, using ATA

[Jim Sturtevant]

I searched the wiki for a solution to allow a user on an
analog ATA to send *81 to block Asterisk CID (or any other * code).

 

The ATA has *81 built in to block the CID the ATA generates.

 

Any examples would be appreciated.

 

 






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[Asterisk-Users] voicemail check for busy message

[Jim Sturtevant]

Is there a way to check to see if a user has recorded a busy
message?  If they haven’t I would prefer to send them to
“u” and play their unavailable greeting.   I know I can
send them to “u” in both cases, but I would like to send them to
“b” if a recorded busy greeting exists.

 

Thanks!






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[Asterisk-Users] cdr from operator initiated calls

[Jim Sturtevant]

I posted a message yesterday that it looked like I was
missing cdr sent to mysql.  I’ve done a little more digging and it
now appears that calls I initiate via TAPI (asttapi) are ending (when the phone
is hungup) with DIALSTATUS=CANCEL

 


 
  
  May 17 08:59:56 VERBOSE[14874]: --
  Lauching Dial(IAX2/sturtevant/18008648331) on IP/200-
  
 
 
  
  May 17 08:59:56 VERBOSE[14874]: --
  Called sturtevant/18008648331
  
 
 
  
  May 17 08:59:56 VERBOSE[14874]: --
  Call accepted by 206.80.70.49 (format ulaw)
  
 
 
  
  May 17 08:59:56 VERBOSE[14874]: --
  Format for call is ulaw
  
 
 
  
  May 17 08:59:56 VERBOSE[14874]: --
  IAX2/sturtevant/2 is making progress passing it to SIP
  
 
 
  
  May 17 08:59:56 DEBUG[14874]: Ooh, voice format changed to
  4
  
 
 
  
  May 17 08:59:56 DEBUG[14874]: Ooh, format changed from
  unknown to ulaw
  
 
 
  
  May 17 08:59:58 DEBUG[14874]: # Testing 66.77.12.74
  with 10.0.0.0
  
 
 
  
  May 17 08:59:58 DEBUG[14874]: Target address 66.77.12.74
  is not local, substituting externip
  
 
 
  
  May 17 09:00:00 DEBUG[14874]: # Testing 66.77.12.74
  with 10.0.0.0
  
 
 
  
  May 17 09:00:00 DEBUG[14874]: Target address 66.77.12.74
  is not local, substituting externip
  
 
 
  
  May 17 09:00:06 WARNING[14874]: Unable to forward frame
  
 
 
  
  May 17 09:00:06 DEBUG[14874]: We're hanging up
  IAX2/sturtevant/2 now...
  
 
 
  
  May 17 09:00:06 VERBOSE[14874]: --
  Hungup 'IAX2/sturtevant/2'
  
 
 
  
  May 17 09:00:06 DEBUG[14874]: Exiting
  with DIALSTATUS=CANCEL.
  
 
 
  
  May 17 09:00:06 DEBUG[14874]: update_user_counter(200) -
  decrement outUse counter
  
 


 

And thus a CDR isn’t being issued?

 

Anyone have any ideas?

 

 






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[Asterisk-Users] mysql debug

[Jim Sturtevant]

I’m cdr_mysql for about a month and it appeared to
be working well.  My server has very little volume 10-20 calls an
hour.  Today there appears to be a gap of about 3 hours where no CDR
records were logged to the MySQL database.  I have done a couple of
reloads during that period which appears to have restarted it as it is now
logging CDR records again without needing to shutdown/restart asterisk or the
server.

 

I reviewed the asterisk logs and don’t see any
mention of problems writing to the database.  Which log should I be
looking at?

 

Is there a way to have * log to both MySQL and a .csv file
(as backup), or should I bite the bullet and move to Radius for capturing
billing records?

 

Thanks

 






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RE: [Asterisk-Users] Interrupting voicemail with "*", dropping to"a" extension. Does it work?

[Jim Sturtevant]

You should set operator=yes in voicemail.conf to get "0" out to work.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Lange
Sent: Thursday, May 12, 2005 7:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Interrupting voicemail with "*", dropping
to"a" extension. Does it work?

On Fri, 2005-05-13 at 12:25 +1000, Adam Goryachev wrote:
> On Thu, 2005-05-12 at 20:44 -0500, John Lange wrote:
> > Very very odd.
> > 
> > Its not a DTMF problem because other tones work fine. # for example
> > skips the OGM as it should.
> > 
> > So could it possible be a config issue?
> > 
> > The voicemail box in question is in the [default] context inside
> > voicemail.conf.
> > 
> > [default]
> > 2048850872 => ,John Lange,[EMAIL PROTECTED]
> > 
> > That tells me I need a this in extensions.conf:
> 
> Does it??
> 
> > [default]
> > exten => a,1,VoicemailMain() ; If they press *, send the user into
VoicemailMain
> > 
> > Am I missing something blazingly obvious?
> 
> I thought that it should be like this:
> [somecontext]
> exten => 1234,1,Voicemail([EMAIL PROTECTED])  <--- Standard check VM
> exten => a,1,VoiceMailMain() <--- press * to get here
> 
> ie, the a extension should be in the same context as the voicemail
> extension, the voicemail context (default) is irrelevant in all this...
> It is ONLY used internally by the voicemail app to determine which
> mailbox this is. Don't confuse them just because they are both called
> context's.

I created this bare-bones example to test it.

[mycontext]
exten => 8761234,1,Voicemail(u2048761234)
exten => a,1,VoiceMailMain() 

It does not work. As mentioned I can skip the OGM by pressing #, but *
(and 0) do nothing.

With verbose set to 9 I see nothing on the console for any of the key
presses.

By the way, I'm using a recent CVS version of Asterisk.

Asterisk CVS-HEAD-05/03/05-16:21:27

This is very baffling.

Are there any other ways of trouble shooting it?

> > I didn't think it could be a config issue because I thought it should at
> > least show the "*" in the console and then complain about no "a"
> > extension or something but I get absolutely nothing in the console.
> 
> Dunno, but try the above... and let us know.
> 
> Regards,
> Adam
-- 
John Lange
President OpenIT ltd. www.Open-IT.ca (204) 885 0872
VoIP, Web services, Linux Consulting, Server Co-Location

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RE: [Asterisk-Users] two questions about the Sipura 841?

[Jim Sturtevant]

What is the purpose of the beeping?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Duffield
Sent: Saturday, May 07, 2005 12:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] two questions about the Sipura 841?

Ok my first question is I have seen messages about a patch for asterisk so
that I can do auto answer on these phones. I found the message in the
archives but I do not have that message as an email still, so I do not have
the attachment. Can anyone tell me where to get it? Also on this phone how
can I set the phone to release the line sooner without playing the anoying
beeping for 5 seconds, I can change how long until the beeping starts but
how do I shorten the beeps?

Thanks

Joel
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005

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RE: [Asterisk-Users] Inexpensive FAX and 800 Number retail service

[Jim Sturtevant]

I've been using www.maxemail.com for quite awhile and they provide great
service.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, May 07, 2005 8:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Inexpensive FAX and 800 Number retail service

Greetings All,

I have a number of projects in the works at the moment and for one of
them, I need to locate an inexpensive and reliable service that can
provide small-office virtual services:

1. FAX to Email
2. Toll Free number with voicemail boxes for Tech Support, Billing
Inquiries, Customer Service, Abuse Reporting, etc...

I have been looking all over the Internet and there seem to be a LOT but I
am at a loss as to which are reliable and cost effective as I have see
rates ranging from $4.95/mo to $29.95/mo.

Being this, I thought that I would ask the experts, which are you guys on
the Asterisk mailing list.

Any help would be greatly appreciated,

Have a great day,
Lonnie


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[Asterisk-Users] end user gui

[Jim Sturtevant]

I’ve reviewed the wiki and other resources and
haven’t been able to locate a tool which would allow an end user to make
changes to their service.  The features and end-user might want to change
is fairly limited (call fwd, number of rings, etc).   This might
require real-time. Thanks in advance.

 






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RE: [Asterisk-Users] Connecting 2 * Together-Pulling hair out

[Jim Sturtevant]

Can you post an example.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Sent: Thursday, May 05, 2005 6:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out

I am not sure what you are trying to do.I have created an IAX2 trunk
between the servers over an internet connection.
Then all you have to do is put in call routing on the trunks to forward the
call to the right place.  Are you using AMP or trying to do it manually.
I found everything a little confusing as well, but it is simple now that I
understand it.


Chris

- Original Message - 
From: "mr. barker" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Sent: Thursday, May 05, 2005 4:43 AM
Subject: [Asterisk-Users] Connecting 2 * Together-Pulling hair out


> 
> 
>  
> 
>   _  
> 
> Subject: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
> 
>  
> 
> I have read the docs on connecting 2* together but am unsure of a few
things
> 
>  
> 
> Do I need a different account for each number that will be called from one
> box to the other ? ie. Do I set up a user account on one and then have the
> other box log into that account when it whats to make a call ?
> 
>  
> 
> I have 2 asterisk boxes and only one of them has the ability to access a
> VoipAccount and PSTN connections.(*box 1). The other holds the SIP
> extensions for the internal SIP users/exten(*box2)
> 
> I would like to be able to have the box with the Sip UA(*box2) on it to be
> able to place a call using the box that has the VoipAccount and PSTN
> connection.  I am able to make multiple UA calls on the VoipAccount and 3
on
> the PSTN lines (only have 3 lines coming in).  I can get it to work if I
> create a user exten on *box1 and map a trunk(which is really only an
exten)
> using the user/password login to that exten from *box2.  However when I
try
> to place a second call when the VOIP line is in use it gives me error (
> basically saying can't use the trunk because it is in use)  I would like
to
> be able to have this exten/trunk to be able to use multiple connections on
> it.
> 
>  
> 
> There must be an easier way to do this I am just not sure how.  I looked
at
> creating IAX trunks but still come up with the Trunk is really an Exten
> name/password .  
> 
>  
> 
> Any help would be appreciated. (my brain is boiling eggs)
> 
>  
> 
> Thank you.
> 
>  
> 
>  
> 
>  
> 
> 






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RE: [Asterisk-Users] Detecting Fax and bad CDRs

[Jim Sturtevant]

I used to own a company which made fax switches and yes your devise must
"answer" the call to detect the fax CNG tone or wait till voicemail answers
and try to detect the CNG tone while the voicemail greeting is being played.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev
Sent: Monday, May 02, 2005 8:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Detecting Fax and bad CDRs

On Mon, 2005-05-02 at 21:21 -0500, Matthew Boehm wrote:
> Here is the current scenario:
> 
> Someone calls my DID. It comes in on PRI. I do "Answer()" then "Wait(1)"
to
> see if there is a fax tone. If there is, goto fax context and do the fax
> thing.
> 
> If no fax tone, then "Dial(SIP/myphone,30)". If I don't answer in the 30
> seconds, goto Voicemail.
> 
> However! If the person were to hangup before the 30 seconds expired, the
> channel would get hung up but the disposition in the CDR (along with the
> billable seconds) would be inaccurate because I "Answer'd" the call
> beforehand.
> 
> Is there a way to change the disposition and the billable seconds to more
> accurately reflect what happened?
> 
> This really is only a big deal for inbound 800 calls; because if I answer
> the call and it is a person and they hang up before voicemail picks up, I
> have to charge the customer for those 10-15 seconds. Plus, I give
customers
> the ability to see their "missed" calls, which is based on a 0 billable
> seconds duration in the CDR.

There are two methods for detecting a fax (that I know of).

1) Allocate a specific DID for your faxes, so any call to that number
will be a fax
2) Answer the call, listen for the fax tone, and if you receive it, then
treat as a fax, else carry on.

AFAIK, you can't listen to the audio prior to answering the call. (You
can sometimes send audio prior to answering the call). So you can't tell
if it is a fax or not.

Personally, I presume you would need to bill your user for that 15
seconds, or else you will end up losing money. You answered the call,
therefore the telco is charging you, if you don't charge your user, then
you will lose.

Of course, I could be wrong, but I doubt it :)

I'd like to know how those 'home/soho' fax machines 'detect' an incoming
fax and allow the inside phone to ring, and only pick up if it is a fax
call.

Regards,
Adam
-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 8304 [EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

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RE: [Asterisk-Users] Asterisk on Radio Tonight

[Jim Sturtevant]

That would be great, thanks


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, April 30, 2005 9:42 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight

If anybody wants it, I recorded the part of the show and will put it up on
my website in about 30 mins. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Jim Sturtevant
|Sent: Sábado, 30 de Abril de 2005 10:53 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight
|
|Kerry, can you put an archive of the audio up on your web site 
|or do we need to record the whole 3hr show.  Also, the 
|schedule on their web site shows 5am EST and other repeats.  
|I'd love to hear the program.
|
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kerry Garrison
|Sent: Saturday, April 30, 2005 2:28 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] Asterisk on Radio Tonight
|
|Kerry Garrison from The Geek Gazette (http://geekgazette.com)  
|will be interviewed tonight on Mick Mick Williams' Cyber Line 
|radio program at 9:00PM PST. The show is broadcast on the USA 
|Radio network. If you do not have a channel in your area, you 
|can listen listen live online 
|<http://www.usaradio.com/listen_live.htm>. The show will cover 
|the basic of what the Asterisk PBX is all about and what it 
|takes to implement a system.
|
|-Kerry
|
|
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|

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RE: [Asterisk-Users] Asterisk on Radio Tonight

[Jim Sturtevant]

Kerry, can you put an archive of the audio up on your web site or do we need
to record the whole 3hr show.  Also, the schedule on their web site shows
5am EST and other repeats.  I'd love to hear the program.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison
Sent: Saturday, April 30, 2005 2:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk on Radio Tonight

Kerry Garrison from The Geek Gazette (http://geekgazette.com)  will be
interviewed tonight on Mick Mick Williams' Cyber Line radio program at
9:00PM PST. The show is broadcast on the USA Radio network. If you do not
have a channel in your area, you can listen listen live online
. The show will cover the basic of
what the Asterisk PBX is all about and what it takes to implement a system.

-Kerry


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RE: [Asterisk-Users] help with compiling addons for cdr

[Jim Sturtevant]

:-) ok... so I feel foolish... Mathew thanks a lot, worked like a charm.

Jim


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Saturday, April 30, 2005 11:16 AM
To: Asterisk Users
Subject: Re: [Asterisk-Users] help with compiling addons for cdr

Well for some reason, you decided to use the stable version of asterisk but
also decided not to use the stable version of addons. Hmm...interesting
decisions.

rm -rf asterisk-addons/
cvs co addons -r v1.0.7

Then it will work.

-Matthew

> From: forums <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sat, 30 Apr 2005 10:24:24 -0700
> To: 
> Subject: [Asterisk-Users] help with compiling addons for cdr
> 
> I'm running Asterisk 1.0.7.  I've checked out using "cvs checkout
> asterisk-addons".
> 
>  
> 
> When I make install I get the following errors:
> 
>  
> 
> app_addon_sql_mysql.c:162:36: macro "AST_LIST_REMOVE" requires 4
arguments,
> but only 3 given
> 
>  
> 
> I'm using the default FC3 mysql:
> 
>  
> 
> mysql-server-3.23.58-16.FC3.1
> 
> perl-DBD-MySQL-2.9003-5
> 
> mysql-3.23.58-16.FC3.1
> 
> mysql-devel-3.23.58-16.FC3.1
> 
> php-mysql-4.3.11-2.4
> 
> libdbi-dbd-mysql-0.6.5-9
> 
> MySQL-python-0.9.2-4
> 
>  
> 
> I've search wiki, etal and have found a couple of references with a
proposed
> patch file, but the patch file fails too.
> 
>  
> 
> I would appreciate any assistance.
> 
>  
> 
>  
> 
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RE: [Asterisk-Users] Asterisk behind NAT

[Jim Sturtevant]

Do you have any phones connected to your * on the internal subnet?  Can they
make outbound calls?  



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Sent: Friday, April 15, 2005 10:15 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

Can you show your outbound peer configuration? If you are registering,
please include that as well.

Thanks

Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 9:44 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

I have...   Externip=x.x.x.xand nothing... Does not seem to help in
anything. Still my provider sees the private IP. 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Sent: Friday, April 15, 2005 9:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

Try setting externip=(asterisk public ip address)

Hth
Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 12:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk behind NAT

Hi List,
I've spent hours researching on this topic, found tons of info, so far it
doesn't work yet.

Here's the scenario

Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to
send calls to an outside provider.

My SIP phones (outside * NAT) are able to register with no problem.


The problem comes when I send a call out to my provider I get no audio in
either way. My provider sees only my private IP.  How can I send my provider
the public IP??


I have in my sip.conf:

[general]
port=5060   ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
nat=yes
externip = x.x.x.x
localnet=192.168.1.0/24

(remember I enabled DMZ on the router so all ports are being forwarded to *)
Despite of what I do, the externip and the localnet my provider only gets my
* private IP.

Any tip how to solve this??

Thanks


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RE: [Asterisk-Users] SPA and NAT traversal

[Jim Sturtevant]

I appreciate everyone's help with setting up an external extension.


Here's a diagram

{SPA2000} - NAT1 - Internet - NAT2 - Asterisk - SIPPhone

SIPPhone is on the same internal subnet as *
NAT2 has a public/staic IP and ports are forwarded to Asterisk

I can successfully do the following:

1. call from SPA2000 - Asterisk VM
2. call from SPA2000 to SIPPhone
3. call from SPA2000 to outside PST as long as I'm using IAX to the ITSP

What I can't do is call from SPA2000 to an outside PSTN if the ITSP is SIP.
When I call the outside phone number I don't hear any ring back or when the
called party answers.   If I have RTP DEBUG on at the CLI I don't see any
RTP packets at all so it appears * is outside the media stream.

SIP-PSTN works great so long as the ITSP is being reached by IAX.

This situation exists regardless of the calue for canreinvite.

SIP PEER:
  * Name   : 202
  Secret   : 
  MD5Secret: 
  Context  : sip
  Language :
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  :
  LastMsgsSent : -1
  Inc. limit   : 0
  Outg. limit  : 0
  Dynamic  : Yes
  Callerid : "" <>
  Expire   : 2749
  Expiry   : 900
  Insecure : no
  Nat  : Always
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr->IP : 24.6.249.xxx Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 202
  Codecs   : 0x4 (ulaw)
  Codec Order  : (ulaw)
  Status   : UNKNOWN
  Useragent: Sipura/SPA2000-2.0.10(c)
  Reg. Contact : sip:[EMAIL PROTECTED]:5060


[202]   ;test ata
type=friend
username=202
secret=
host=dynamic
nat=no
reinvite=no ;stay in the call
canreinvite=no  ;keeps asterisk in the media stream
disallow=all
allow=ulaw
context=sip

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Saturday, April 09, 2005 1:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SPA and NAT traversal

> In your second option using a STUN server would I need to setup my
> own STUN server? 

No, use FWD or xten's STUN servers.

-- 
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698

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RE: [Asterisk-Users] SPA and NAT traversal

[Jim Sturtevant]

In your second option using a STUN server would I need to setup my own STUN
server?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Saturday, April 09, 2005 12:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SPA and NAT traversal

> Thank you for your reply.  There is a wealth of information on the
> wiki, etc.   I turned on RTP debug and the SPA is not sending it's
> public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP
> packets are going nowhere... 

Do I understand your question correctly:

You have an SPA behind NAT1 and * and a second SIP device behind NAT2. Both
devices register, but calls between the devices result in no audio?

If that is the case, you can do one of two things:

- set canreinvite=no for the devices' sip.conf entries, or
- teach both devices to *stop* using their internal IPs for all
communications and remove nat=yes from the entry for the SIP device inside
NAT2.

To set the SPA to give the correct IP, enable STUN, add a STUN server, and
say Yes to "Substitue VIA Addr".

-- 
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698

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RE: [Asterisk-Users] SPA and NAT traversal

[Jim Sturtevant]

Thank you for your reply.  There is a wealth of information on the wiki,
etc.   I turned on RTP debug and the SPA is not sending it's public IP it is
sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere...


The SPA is behind a NAT and traversing the public IP network to get to the *
server.  It is successfully registering, thus I can ring a phone registered
locally to the * server.

I made sure localnet=192.168.2.9/255.255.255.0 (my local cfg for *)  and
externip=65.87.x.x (which is the public IP of my * server).  The * server is
behind a NAT as well with the 5060 and 16384-32767 UDP ports open.  

Based on RTP debug it appears the RTP packets are making it to the * server,
the problem is the return address is the internal NAT address of the SPA
192.168.1.100 and not it's public address.

Are you willing to share your Martha collection or are you going to keep it
to yourself? :-)


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Saturday, April 09, 2005 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SPA and NAT traversal

Jim Sturtevant wrote:

> I was hoping someone might help me diagnose a NAT issue with an SPA-2000
and
> my * server.  
> 
> My SPA is behind a NAT accessing a server which is also behind a NAT but
SIP
> and RTP ports are forwarded to it.
> 
> My SPA can successfully register.  It can call another extension which is
> inside the * local net and the inside phone can call the SPA.  But, no
> speech path either way.  I have NAT=YES and the two invite parameters are
> set to NO.

I'm desperately trying to get your sip.conf file telepathically but 
all I'm getting is images from your Martha Stewart porn collection. 
*shudder*

In addition to nat=yes you also need localnet= and externip= set, as 
shown in sip.conf.sample.


-- 
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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[Asterisk-Users] SPA and NAT traversal

[Jim Sturtevant]

I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and
my * server.  

My SPA is behind a NAT accessing a server which is also behind a NAT but SIP
and RTP ports are forwarded to it.

My SPA can successfully register.  It can call another extension which is
inside the * local net and the inside phone can call the SPA.  But, no
speech path either way.  I have NAT=YES and the two invite parameters are
set to NO.

Thoughts?


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RE: [Asterisk-Users] Buying some Polycom IP300s

[Jim Sturtevant]

I’ve been using the IP 500 and like
it a great deal.  Be aware that the IP300 does not have a speaker phone.

 








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RE: [Asterisk-Users] username/password for PolyCom IP500 web interface?

[Jim Sturtevant]

http://www.voip-info.org/wiki-Polycom+Phones

It's in the admin guide.  User: Polycom; password: 456

Good luck.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Garrett Nelson
Sent: Wednesday, March 30, 2005 7:24 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] username/password for PolyCom IP500 web interface?

Ok, I am still working on getting this PolyCom phone working with Asterisk.
I have been looking all over, but I have not been able to find the username
and password for the web interface on this phone.

I found some site that said it was Polycom and spip, but that does not work.
Anyone else have any ideas what it might be? Both PolyCom and the place I
bought the phone from are useless for support.

-Garrett

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[Asterisk-Users] voicemail sending blank .WAV file via email

[Jim Sturtevant]

I’ve recently installed asterisk and am working with
the email a voicemail function.  When a voice msg is left 4 files are
created in the /var/spool… directory.  They are .gsm, .txt, .wav and
.WAV.  The .wav (lower case) has the actual audio in it, the .WAV is a
short blank audio file.  When * emails the message it is sending the .WAV
and not the .wav file.  Any thoughts would be appreciated.

 

Jim

 






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RE: [Asterisk-Users] small qos switch

[Jim Sturtevant]

What product from Sangom and at what price point?  Thx

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of steve szmidt
Sent: Sunday, March 27, 2005 6:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] small qos switch

On Sunday 27 March 2005 13:48, Jim Sturtevant wrote:
> How about considering the linksys WRT54G (approx $59) with SVEASOFT
> firmware ($29) www.sveasoft.com which provides QOS by port, IP, and/or
> traffic type plus VPN, SNMP, etc.  and WiFi to boot.
>

Maybe because the Sangoma card will run circles around it purely from a 
performance view, never mind the quality. Which is usually important to a 
business...

>
> > You can buy 400 series servers from Dell for around $350, new.  Run your
> > firewall (iptables) and NAT on that computer.  You can get a Sangoma DSL
> > PCI card for about $115--it has QoS.  You'll have professional grade
> > infrastructure for not that much money.  What's not elegant about that?

-- 

Steve Szmidt

"They that would give up essential liberty for temporary safety 
deserve neither liberty nor safety."
Benjamin Franklin
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RE: [Asterisk-Users] small qos switch

[Jim Sturtevant]

How about considering the linksys WRT54G (approx $59) with SVEASOFT firmware
($29) www.sveasoft.com which provides QOS by port, IP, and/or traffic type
plus VPN, SNMP, etc.  and WiFi to boot.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter
Sent: Sunday, March 27, 2005 10:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] small qos switch


> 
> You could also get an old, cheap computer off eBay put it between the
> switch(es) and the dsl modem, install linux and then use it to do your
> QoS prioritization.  Not very elegant or professional looking, but it
> would work if you don't care about such niceties.
> 
> 
> 
You can buy 400 series servers from Dell for around $350, new.  Run your 
firewall (iptables) and NAT on that computer.  You can get a Sangoma DSL 
PCI card for about $115--it has QoS.  You'll have professional grade 
infrastructure for not that much money.  What's not elegant about that?

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[Asterisk-Users] ata vs digium card

[Jim Sturtevant]

What are the advantages of the Digium PCI cards for FXS ports
vs standalone ATAs?






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[Asterisk-Users] voicemail2

[Jim Sturtevant]

Can anyone point me to information on "the advanced voicemail system" (aka
voicemail2).  I saw a note that it is not yet part of the standard
distribution.  What are the features and when/if is it likely to be added to
the standard dist.

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[Asterisk-Users] newbe: help with registration

[Jim Sturtevant]

I would appreciate any thoughts anyone
might have on configuring this registration.  Thanks

 

I have a SIP account that I can successfully register with
XTEN and a Sipura-2000.  I have yet to be able to get it to authorize with
*.

 

 

My XTEN looks like:

Username:   
001234

Password:


Authorization
Username:   
001234

Domain:   
domain.net

 

Register with
domain:
yes

Use as Outbound
proxy:
yes

Manual Override
host:
host.net

 

What I’ve determined is that the SIP server (host.net)
needs to see [EMAIL PROTECTED].

 

From XTEN successful registration:  

Authorization: Digest
username="001234",realm="BroadWorks",nonce="254549745",uri="sip:domain.net",response="19107e21c530aa0d3efbda6778a3c41e",algorithm=MD5

 

From * failed registration

Authorization: Digest
username="[EMAIL PROTECTED]", realm="BroadWorks",
algorithm=MD5, uri="sip:host.net", nonce="254952151",
response="7a39a5a3a774c4225c5bab8858a1fbf2", opaque=""

 

The URI token is different so I get a SIP/2.0
401 Unauthorized from host.net

 

SIP.CONF

register
=> [EMAIL PROTECTED]::[EMAIL PROTECTED]@host.net/210

 

[cp01]

type=friend

username=001234

secret=Xxxx

host=host.net

insecure=very

canreinvite=no

 

 

 

 

ASTERISK CONSOLE LOG

 



09:49:44.2 

REGISTER sip:host.net SIP/2.0

To: new test

From: new
test;tag=6b1a582b

Via: SIP/2.0/UDP
192.168.2.2:8324;branch=z9hG4bK-d87543-741421617-1--d87543-;rport

Call-ID: 29692c204a75e445

CSeq: 6 REGISTER

Contact: 

Expires: 3600

Max-Forwards: 70

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

User-Agent: eyeBeam release 3003x stamp 16296
(sn:0ea18597a34ba5ef8f48)

Authorization: Digest username="001234",realm="BroadWorks",nonce="254549745",uri="sip:host.net",response="19107e21c530aa0d3efbda6778a3c41e",algorithm=MD5

Content-Length: 0

 

 

09:49:44.3 

RECEIVING FROM: 66.151.54.132:5060

SIP/2.0 200 OK

To: new test;tag=SD1ode099-

From: new
test;tag=6b1a582b

Via: SIP/2.0/UDP
192.168.2.2:8324;received=65.87.19.106;branch=z9hG4bK-d87543-741421617-1--d87543-;rport=8324

Call-ID: 29692c204a75e445

CSeq: 6 REGISTER

Contact:
;expires=30

Contact:
;expires=29

Content-Length: 0



 






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[Asterisk-Users] help with registration

[Jim Sturtevant]

I have a SIP account that I can successfully register with
XTEN and a Sipura-2000.  I have yet to be able to get it to authorize with
*.

 

 

My XTEN looks like:

Username:    001234

Password: 

Authorization Username:    001234

Domain:    domain.net

 

Register with domain: yes

Use as Outbound proxy: yes

Manual Override host: host.net

 

What I’ve determined is that the SIP server (host.net)
needs to see [EMAIL PROTECTED].

 

From XTEN successful registration:  

Authorization: Digest username="001234",realm="BroadWorks",nonce="254549745",uri="sip:domain.net",response="19107e21c530aa0d3efbda6778a3c41e",algorithm=MD5

 

From * failed registration

Authorization: Digest username="[EMAIL PROTECTED]",
realm="BroadWorks", algorithm=MD5, uri="sip:host.net",
nonce="254952151",
response="7a39a5a3a774c4225c5bab8858a1fbf2", opaque=""

 

The URI token is different so I get a SIP/2.0
401 Unauthorized from host.net

 

SIP.CONF

register
=> [EMAIL PROTECTED]::[EMAIL PROTECTED]@host.net/210

 

[cp01]

type=friend

username=001234

secret=Xxxx

host=host.net

insecure=very

canreinvite=no

 

 

 

 

ASTERISK CONSOLE LOG

 



09:49:44.2


REGISTER
sip:host.net SIP/2.0

To:
new test

From:
new test;tag=6b1a582b

Via:
SIP/2.0/UDP 192.168.2.2:8324;branch=z9hG4bK-d87543-741421617-1--d87543-;rport

Call-ID:
29692c204a75e445

CSeq:
6 REGISTER

Contact:


Expires:
3600

Max-Forwards:
70

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

User-Agent:
eyeBeam release 3003x stamp 16296 (sn:0ea18597a34ba5ef8f48)

Authorization:
Digest username="001234",realm="BroadWorks",nonce="254549745",uri="sip:host.net",response="19107e21c530aa0d3efbda6778a3c41e",algorithm=MD5

Content-Length:
0

 

 

09:49:44.3


RECEIVING
FROM: 66.151.54.132:5060

SIP/2.0 200 OK

To:
new test;tag=SD1ode099-

From:
new test;tag=6b1a582b

Via:
SIP/2.0/UDP 192.168.2.2:8324;received=65.87.19.106;branch=z9hG4bK-d87543-741421617-1--d87543-;rport=8324

Call-ID:
29692c204a75e445

CSeq:
6 REGISTER

Contact:
;expires=30

Contact:
;expires=29

Content-Length:
0



 






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[Asterisk-Users] trying to get trunk to register with * behind NAT

[Jim Sturtevant]

i've got * and phones in small home network all behind NAT. Outbound to
iconnect proxy works great. Now to get in/out working with another carrier.
Carrier2, Commpartners, i have working with one of the phones and a soft phone
without * just fine.  
 
Next I register the phone with * fine. Create a trunk, but it the trunk fails
to register... help 
 
I'm getting the following msg during registration: 
to 66.218.79.147:5060 
Retransmitting #5 (no NAT): 
REGISTER sip:x.net SIP/2.0 
 
SIP.CONF= 
; Note: If your SIP devices are behind a NAT and your Asterisk 
; server isn't, try adding "nat=1" to each peer definition to 
; solve translation problems. 
 
[general] 
 
; 2005-03-14 
externip={external ip} 
localnet=192.168.2.0 ;(i.e.192.168.1.0)  
localmask=255.255.255.0 ;(or whatever your localnet mask)  
nat=yes  
 
port = 5060 ; Port to bind to (SIP is 5060) 
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) 
disallow=all 
allow=ulaw 
allow=alaw 
context = from-sip-external ; Send unknown SIP callers to this context 
callerid = Unknown 
 
#include sip_nat.conf 
#include sip_additional.conf 
 
 
register={userid}:[EMAIL PROTECTED] 
 
[200] 
username=200 
type=friend 
secret=??? 
qualify=no 
port=5060 
pickupgroup= 
nat=never 
mailbox= 
host=dynamic 
dtmfmode=rfc2833 
disallow= 
context=from-internal 
canreinvite=no 
callgroup= 
callerid="Demo" <200> 
allow= 
 
[CP 9929] 
username={username} 
type=peer 
secret={password} 
host={host} 
 
[cp-in-01] 
type=user 
context=from-pstn 






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