Re: [asterisk-users] Faxing over Carrier SIP trunk/g711 ?

[Lee Howard]

Jason Aarons (US) wrote:
>
> Anyone have a customer sending/receiving multi-page faxes over Verizon 
> Business SIP trunk/g711 ?
>
> Verizon Business indicates they don’t support it, and I have 2 recent 
> customers that it doesn’t work for, and 1 current large customer 
> telling me he’s going to make it work .
>
> The issues is the latency/jitter on fax/g711 over Verizon Business 
> seems to spit out only 11 pages of a 15 page fax.
>
> Anyone having faxing over PSTN SIP over G711 that is working? Any advice?
>

Read: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf

Thanks,

Lee.

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Re: [asterisk-users] MeetMe Options Enter Leave Sound

[D Tucny]

2009/7/24 Stefan Schmidt 

> Hello,
>
> i´ve a question about the Meetme Options. How could i play a enter and
> leave sound but without recording the user name first. I just want a
> User joined conferenc and a user leaved.
>
> With the i or I Option i have to record the name first.
>
> Is there any way of doing this? As i can see in the Meetme help the
> background agi couldnt be used on non dahdi channel which i will have on
> this server cause there is no direct Pri link, just SIP.
>

By default join and leave sounds are played when someone joins and leaves
the conference...

d
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[asterisk-users] User Authentication in sip.conf

[velusamy velu]

Dear all,
 I want to setup the incoming calls, that don't use authentication in
sip.conf file.
 My configurations as follows,

[2000]
type=peer
host=dynamic
insecure=port,invite; (both)
context=Testing

But when I call '2000', I noticed the following message in Asterisk console,

"NOTICE[5702]: chan_sip.c:10543 handle_request_invite: Failed to
authenticate user "Velusamy" 
>;tag=yj66acQcycvrN"

What would be the problem??

Please help me to solve this problem.

Best Regards,
Velusamy
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Re: [asterisk-users] Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1, and 1.6.2.0-beta4 Release Announcement

[Klaverstyn, David C]

Faxing over SIP never worked for me.  The faxes would always fail.  When
I saw the information about T.38, I decided to immediately upgrade to
1.6.0.11-rc2 from 1.6.0.10 and try it.

I was amazed.  Without having to change anything in my configuration
faxes just worked.  I have tested it with multiple faxes, short and
long, and faxes with images and they all came through.

Well done guys.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Team
Sent: Monday, 3 August 2009 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6.0.11-rc2, 1.6.1.2,
1.6.1.3-rc1,and 1.6.2.0-beta4 Release Announcement

The Asterisk Development Team is pleased to announce the the second
release 
candidate of 1.6.0.11, the release of 1.6.1.2, the first release
candidate of
1.6.1.3, and the fourth beta of 1.6.2.0.  These releases are available
for
immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/ .

The release of 1.6.1.2 fixes a remote crash security vulnerability in
the RTP
stack.  The related security advisory AST-2009-004 has been released
along
with this announcement.  Please read that advisory for more information.

The release candidates and betas, in addition to other fixes, contain a
major
re-work of the T.38 support in Asterisk.  If you've been having trouble
with
T.38 in the 1.6 series, you are strongly encouraged to try one of these
release candidates to determine if these changes fixed your T.38 issues.

For a full list of changes in these releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-
1.6.0.11-rc2
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-
1.6.1.2
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-
1.6.1.3-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-
1.6.2.0-beta4

Thank you for your continued support of Asterisk!

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[asterisk-users] AST-2009-004: Remote Crash Vulnerability in RTP stack

[Asterisk Security Team]

   Asterisk Project Security Advisory - AST-2009-004

   ++
   |   Product| Asterisk|
   |--+-|
   |   Summary| Remote Crash Vulnerability in RTP stack |
   |--+-|
   |  Nature of Advisory  | Exploitable Crash   |
   |--+-|
   |Susceptibility| Remote unauthenticated sessions |
   |--+-|
   |   Severity   | Critical|
   |--+-|
   |Exploits Known| No  |
   |--+-|
   | Reported On  | July 27, 2009   |
   |--+-|
   | Reported By  | Marcus Hunger |
   |--+-|
   |  Posted On   | August 2, 2009  |
   |--+-|
   |   Last Updated On| August 2, 2009  |
   |--+-|
   |   Advisory Contact   | Mark Michelson|
   |--+-|
   |   CVE Name   | |
   ++

   ++
   | Description | An attacker can cause Asterisk to crash remotely by  |
   | | sending malformed RTP text frames. While the attacker|
   | | can cause Asterisk to crash, he cannot execute arbitrary |
   | | remote code with this exploit.   |
   ++

   ++
   | Resolution | Users should upgrade to a version listed in the   |
   || "Corrected In" section below. |
   ++

   ++
   |   Affected Versions|
   ||
   |Product| Release Series |   |
   |---++---|
   | Asterisk Open Source  | 1.2.x  | Unaffected|
   |---++---|
   | Asterisk Open Source  | 1.4.x  | Unaffected|
   |---++---|
   | Asterisk Open Source  | 1.6.x  | All 1.6.1 versions|
   |---++---|
   |Asterisk Addons| 1.2.x  | Unaffected|
   |---++---|
   |Asterisk Addons| 1.4.x  | Unaffected|
   |---++---|
   |Asterisk Addons| 1.6.x  | Unaffected|
   |---++---|
   |   Asterisk Business Edition   | A.x.x  | Unaffected|
   |---++---|
   |   Asterisk Business Edition   | B.x.x  | Unaffected|
   |---++---|
   |   Asterisk Business Edition   | C.x.x  | Unaffected|
   |---++---|
   |  AsteriskNOW  |  1.5   | Unaffected|
   |---++---|
   |  s800i (Asterisk Appliance)   | 1.2.x  | Unaffected|
   ++

   ++
   |  Corrected In 

Re: [asterisk-users] asterisk 1.6 call forwarding

[D Tucny]

2009/7/31 pepesz76 

> Dear All,
>
> I'n trying to make a simple call forwarding, however I have small
> problem when evaluating an expresion.
>
> Here is my extensions.conf
> ...
>
>
> ; Unconditional Call Forward
> exten => _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
> exten => _#21*X.,2,Hangup()
> exten => #21#,1,Set(ignored=${DB_DELETE(CFIM/${CALLERID(num)})})
> exten => #21#,2,Hangup()
> ...
> exten => 50,1,Set(CFIM=${DB(CFIM/${EXTEN})})
> exten => 50,n,GotoIf($[${CFIM}=""]?start)  ;<- THIS IS WRONG, but not
> sure what should it look like?
> exten => 50,n,Dial(SIP/${CFIM},30)
> exten => 50,n,Dial(SIP/${EXTEN},30
> ...
>
> First part properly sets and deletes string in database
> Second part works as the forwarding is set, however if it is not set
> then CFIM is empty and I got:
> WARNING[9752]: ast_expr2.fl:434 ast_yyerror: ast_yyerror():  syntax error:
> syntax error, unexpected '=', expecting $end; Input:
> =''
>
> Can someone suggest the solution?
>
>
What it's doing in this case is first extracting the variable value then
performing the evaluation of the condition, so, when there is no value set,
what it's effectivly trying in the condition is $[=""], hence the unexpected
'=' message. If you quote the variable, i.e. $["${CFIM}"=""] then in the
event the variable is unset or contains an empty value the condition will
still be at least $[""=""] and it should work...

d
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[asterisk-users] Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1, and 1.6.2.0-beta4 Release Announcement

[Asterisk Team]

The Asterisk Development Team is pleased to announce the the second release 
candidate of 1.6.0.11, the release of 1.6.1.2, the first release candidate of
1.6.1.3, and the fourth beta of 1.6.2.0.  These releases are available for
immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ .

The release of 1.6.1.2 fixes a remote crash security vulnerability in the RTP
stack.  The related security advisory AST-2009-004 has been released along
with this announcement.  Please read that advisory for more information.

The release candidates and betas, in addition to other fixes, contain a major
re-work of the T.38 support in Asterisk.  If you've been having trouble with
T.38 in the 1.6 series, you are strongly encouraged to try one of these
release candidates to determine if these changes fixed your T.38 issues.

For a full list of changes in these releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.0.11-rc2
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.2
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.3-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.0-beta4

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] T.38 and reinvite

[David Backeberg]

On Sun, Aug 2, 2009 at 3:30 PM, Benny Amorsen wrote:
> Is there a way to get ONLY T.38 reinvite without Asterisk trying to get
> out of the media path?

That's an excellent question. As you've realized, T.38 works by
initializing the SIP connection as audio over a chosen codec, and then
if T.38 is offered there's a reinvite to initiate the T.38 session.

Because of this behavior, and because you want to disable the reinvite
for normal audio calls, I think you will have a to use a separate SIP
trunk for faxing. I can't think of any other way to do it. So a second
parallel SIP trunk for each location sounds like the your way out of
this problem.

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Re: [asterisk-users] Modem

[Cary Fitch]

"Yes, they changed their name to Copaco for Compania Paraguaya de
Comunicaciones. It's basically the same company ruling the whole country.
":

Oh, like AT&T and Verizon here. :-(

 

Please pardon the editorial comment, list.

Cary Fitch

 

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Re: [asterisk-users] Modem

[Carlos Ruiz Diaz]

You understand perfectly fine the situation :) . I'm not saying that
Paraguay has the worse economy in South-America, but we need to work much
harder to get latest technology or to mount a tiny/small laboratory.

You will get amized if you see the things that we have done with pieces of
hardware considered as garbage in USA :D

>  Does Antelco still dominate the telco
market in Paraguay, I wonder.

Yes, they changed their name to Copaco for Compania Paraguaya de
Comunicaciones. It's basically the same company ruling the whole country. :S


Thanks to all for answering my question.

On Sun, Aug 2, 2009 at 4:56 PM, Jared Smith  wrote:

> On Sun, 2009-08-02 at 14:54 -0400, Carlos Ruiz Diaz wrote:
> > Why  PC modems were not used as FXO devices? Why chan_modem was
> > deprecated? it seemed a nicer option instead of buying expensive
> > gateways.
>
> This question has been answered many times, but just for the fun of it
> I'll answer it again:
>
> If PC modems had been ideal telephony cards, we'd still be using them.
>
> My own experience with using modems as FXO devices (long before I became
> a Digium employee) was that they were awful.  I encountered problems
> with echo, half-duplex audio, and lack of far-end disconnect
> supervision.  All of those problems are solved with most modern
> telelphony cards (except for the ultra-cheap cards, which are still just
> modems).  To put it frankly, I wouldn't wish one of those modems on my
> worst enemies.
>
> > Anyway, for people living really far from USA the price gets
> > incremented twice or more and this is without considering the
> > conversion between currencies.
> >
> > 1 $ = 5100 Gs., not cheap at all.
>
> I understand that the cards are disproportionately expensive in many
> parts of the world as compared to the United States, because of the
> difference in economies. I spent a couple of years in Paraguay in the
> mid 90s, and know what it's like to pay outrageous prices for
> specialized electronics just because they have to be imported from other
> countries. (I'm guessing that you're from Paraguay, based on on the
> monetary conversion you gave.  Does Antelco still dominate the telco
> market in Paraguay, I wonder?)
>
> That being said, the cost per port of the Digium cards (or any of our
> competitors who design their own cards) is still much lower than what
> you'd pay for traditional telephony cards, such as those manufactured by
> Dialogic or Aculab.
>
> I know that probably doesn't help you afford to be able to buy a more
> expensive card, but hopefully you have a better understanding of why we
> don't use modems as FXO devices.  If your time and sanity are worth
> anything at all, it's a worthwhile investment to buy a good solid
> telephony card.
>
> --
> Jared Smith
> Training Manager
> Digium, Inc.
>
>
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Re: [asterisk-users] Asterisk and E1 Cards

[Tarek Sawah]

do you suggest buying a licensed Software from Digium? 


Date: Sun, 2 Aug 2009 18:53:16 -0400
From: stot...@asteriskhelpdesk.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk and E1 Cards



On Sun, Aug 2, 2009 at 6:37 PM, Tarek Sawah  wrote:






Greetings List,

Greetings
 i have a new question regarding Asterisk and E1 Cards

a client of mine is requiring an Asterisk Server with 2 E1s.
the scenario is the following
they want 400 extensions to register with the system.. and required 64 
concurrent calls. 

Unless I am mistaking, or you are including internal calls, 2 E1 would handle 
62 or 60 if PRI.  400 extensions should be no problem.  On the same LAN?

 added to it that they are expecting the system to have an IVR to do some DB 
querying.

the setup I have in mind is a Core2duo Server with 3 GB Ram and a Raid0 and a 
TE220B card.
Hard to say which would be better, two lower spec (cheaper) boxen setup 
identically, one as a cold swap.  Backup DB, conf, and whatever, nightly.  I 
have done this for many customers.


RAID 0 is basically useless for Asterisk and sets yourself up for double chance 
of disk failure.  RAID 1 is the way to go.
 

we have not faced this need from a client as we usually provide SIP Services 
only.. so my questions are the following
1- how many calls my setup will be able to handle? and if it won't handle 2 E1s 
what is the best server i can get for that?

You can handle that easily unless you are doing heavy codecs like G729 or 
recording every call.
 

2- E1 supports Ulaw and Alaw codecs so we won't be needing G729 nor G723 
encoding and decoding? or we will have to use such codecs? (I'm concirned about 
the System resources)


You should have said that first ;)  A pentium 4 2.8ghz could handle this 
without breaking a sweat.
 
Thank you in Advance for your help and support.
regards
Tarek



-- 
Thanks,
Steve Totaro 
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] Asterisk and E1 Cards

[Steve Totaro]

On Sun, Aug 2, 2009 at 6:37 PM, Tarek Sawah  wrote:

>  Greetings List,
>

Greetings


> i have a new question regarding Asterisk and E1 Cards
> a client of mine is requiring an Asterisk Server with 2 E1s.
> the scenario is the following
> they want 400 extensions to register with the system.. and required 64
> concurrent calls.
>

Unless I am mistaking, or you are including internal calls, 2 E1 would
handle 62 or 60 if PRI.  400 extensions should be no problem.  On the same
LAN?


> added to it that they are expecting the system to have an IVR to do some DB
> querying.
> the setup I have in mind is a Core2duo Server with 3 GB Ram and a Raid0 and
> a TE220B card.
>

Hard to say which would be better, two lower spec (cheaper) boxen setup
identically, one as a cold swap.  Backup DB, conf, and whatever, nightly.  I
have done this for many customers.

RAID 0 is basically useless for Asterisk and sets yourself up for double
chance of disk failure.  RAID 1 is the way to go.


>
> we have not faced this need from a client as we usually provide SIP
> Services only.. so my questions are the following
> 1- how many calls my setup will be able to handle? and if it won't handle 2
> E1s what is the best server i can get for that?
>

You can handle that easily unless you are doing heavy codecs like G729 or
recording every call.


>
> 2- E1 supports Ulaw and Alaw codecs so we won't be needing G729 nor G723
> encoding and decoding? or we will have to use such codecs? (I'm concirned
> about the System resources)
>

You should have said that first ;)  A pentium 4 2.8ghz could handle this
without breaking a sweat.


> Thank you in Advance for your help and support.
> regards
> Tarek
>
>


-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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[asterisk-users] Asterisk and E1 Cards

[Tarek Sawah]

Greetings List,
i have a new question regarding Asterisk and E1 Cards
a client of mine is requiring an Asterisk Server with 2 E1s.
the scenario is the following
they want 400 extensions to register with the system.. and required 64 
concurrent calls. 
added to it that they are expecting the system to have an IVR to do some DB 
querying.
the setup I have in mind is a Core2duo Server with 3 GB Ram and a Raid0 and a 
TE220B card.
we have not faced this need from a client as we usually provide SIP Services 
only.. so my questions are the following
1- how many calls my setup will be able to handle? and if it won't handle 2 E1s 
what is the best server i can get for that?
2- E1 supports Ulaw and Alaw codecs so we won't be needing G729 nor G723 
encoding and decoding? or we will have to use such codecs? (I'm concirned about 
the System resources)
Thank you in Advance for your help and support.
regards
Tarek

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[asterisk-users] AstLinux 0.6.7 released

[Darrick Hartman]

The Astlinux Development Team is happy to announce the release of 
AstLinux 0.6.7.  This release is a security and bugfix release with no 
new features.  All current users of AstLinux are encouraged to upgrade.

Current users can upgrade either from the web interface or by issuing 
the following commands from the CLI.

upgrade-run-image check http://mirror.astlinux.org/firmrware

(which should report ... Newest available version is: astlinux-0.6.7)

then

upgrade-run-image upgrade http://mirror.astlinux.org/firmware

New users can download the installation images or ISO images from the 
Sourceforge project site.  Note that sourceforge made some changes so 
not all 0.6.7 versions appear under 'newest files'.  You'll need to 
browse to the bottom of the files page to the 0.6.7

https://sourceforge.net/projects/astlinux/

Release notes are available here:

http://www.astlinux.org/releasenotes/0.6.7

Regards,

The Astlinux Development Team

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Re: [asterisk-users] Modem

[Tzafrir Cohen]

On Sun, Aug 02, 2009 at 03:07:08PM -0400, jon pounder wrote:
> Carlos Ruiz Diaz wrote:
> > Hello list,
> >
> > Why  PC modems were not used as FXO devices? Why chan_modem was 
> > deprecated? it seemed a nicer option instead of buying expensive gateways.

Because nobody bothered writing drivers for any of them.

> 
> the digium single fxo cards and clones for about $10 ARE modems.
> you can get a sip gateway fxo + fxs in one box for about $50

At the time they were. Nowadays they cost more. Those specific modems
are now obsolete and not manufactured anymore.

-- 
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+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] how to sniff RTP and SIP traffic only

[Timothy Weidner]

To make your life a little easier, you can use the following filter:
sip or sdp or rtp

Just insert that into the filter query field in wireshark and it'll show you
what you need.

On Sun, Aug 2, 2009 at 12:49 PM, Joe Carroll  wrote:

>  Wireshark will support this…
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Xavier Cardil
> *Sent:* Monday, June 29, 2009 5:51 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] how to sniff RTP and SIP traffic only
>
>
>
> Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster
> debugging ?
>
> Thanks.
>
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Re: [asterisk-users] Modem

[Jared Smith]

On Sun, 2009-08-02 at 14:54 -0400, Carlos Ruiz Diaz wrote:
> Why  PC modems were not used as FXO devices? Why chan_modem was
> deprecated? it seemed a nicer option instead of buying expensive
> gateways.

This question has been answered many times, but just for the fun of it
I'll answer it again:

If PC modems had been ideal telephony cards, we'd still be using them.

My own experience with using modems as FXO devices (long before I became
a Digium employee) was that they were awful.  I encountered problems
with echo, half-duplex audio, and lack of far-end disconnect
supervision.  All of those problems are solved with most modern
telelphony cards (except for the ultra-cheap cards, which are still just
modems).  To put it frankly, I wouldn't wish one of those modems on my
worst enemies.

> Anyway, for people living really far from USA the price gets
> incremented twice or more and this is without considering the
> conversion between currencies. 
> 
> 1 $ = 5100 Gs., not cheap at all.

I understand that the cards are disproportionately expensive in many
parts of the world as compared to the United States, because of the
difference in economies. I spent a couple of years in Paraguay in the
mid 90s, and know what it's like to pay outrageous prices for
specialized electronics just because they have to be imported from other
countries. (I'm guessing that you're from Paraguay, based on on the
monetary conversion you gave.  Does Antelco still dominate the telco
market in Paraguay, I wonder?)

That being said, the cost per port of the Digium cards (or any of our
competitors who design their own cards) is still much lower than what
you'd pay for traditional telephony cards, such as those manufactured by
Dialogic or Aculab.

I know that probably doesn't help you afford to be able to buy a more
expensive card, but hopefully you have a better understanding of why we
don't use modems as FXO devices.  If your time and sanity are worth
anything at all, it's a worthwhile investment to buy a good solid
telephony card.

-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] Voicemail Error

[Ron]

Sorry i think i forgot to mention that i have /var/spool/asterisk as a 
directory from another server mounted via sshfs. when i don't use a 
remote directory recording works fine. not sure if this is a permission, 
but i switched to user asterisk and created new files on the remote 
directory i can create and overwrite files. any ideas? TIA

Regards
Ron

Ron wrote:
> Hi All,
> 
> I'm trying to test asterisk voicemail on recording my own unavailable 
> message, busy message or temporary message. I was looking at the console 
> and saw this message:
> 
> app_voicemail store_file Memory map failed
> 
> Then i looked at /var/spool/asterisk/ there were no recorded 
> greetings. what does the error mean? TIA
> 
> Regards
> Ron
> 
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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Tim Panton]

I don't know then. My understanding is that the message is caused by
the wrong skypeforasterisk process running.

- did you (ever) run it as a different user ?

If it is a test box, you could try a full reboot.

Tim.

On 2 Aug 2009, at 19:35, Emrah wrote:


Hi Tim,

I don't have any skypeforasterisk process  running. I tried to killall
-9 asterisk but it did not solve my issue.
Any other suggestions?
Thanks for your help,
Emrah
Tim Panton wrote:

I had that too, I cured it by kill -9 'ing the skypeforasterisk
process that was left over from
the previous version of the beta.

Hope that helps.

Tim.

On 2 Aug 2009, at 11:20, Emrah wrote:


I reported an issue on Mantis (#14).
Waiting for an update.
http://betareports.digium.com/mantis/view.php?id=14

Emrah
Emrah wrote:

Hi Thomas,

I am experiencing the same problem, with the same error messages.
Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686

Regards,
Emrah
Thomas Kenyon wrote:


Thomas Kenyon wrote:


[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ:  
bad

magic number 0x25765ca0 for 0x1390e20
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ:  
bad

magic number 0x25765ca0 for 0x1390e20




chef*CLI> skype show users
Skype Users
[2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ:  
bad

magic number 0x70796b73 for 0x7f4fe0044340
[2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ:  
bad

magic number 0x70796b73 for 0x7f4fe0044340

Sorry, these are the error messages.

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[asterisk-users] T.38 and reinvite

[Benny Amorsen]

I have a setup with a number of customer Asterisks with T.38 enabled.
This works quite well for each customer sending faxes between branch
offices. 

They all have a SIP trunk to a central Asterisk, which connects them to
the PSTN through various providers on dedicated lines. I cannot enable
reinvite on those SIP trunks, because that would allow calls from the
customer's phones to get reinvited and talk directly to the central
Asterisk -- and there are firewall rules forbidding that.

Is there a way to get ONLY T.38 reinvite without Asterisk trying to get
out of the media path?


/Benny



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Re: [asterisk-users] Modem

[jon pounder]

Carlos Ruiz Diaz wrote:
> I did not know that the price was that low. Anyway, for people living 
> really far from USA the price gets incremented twice or more and this 
> is without considering the conversion between currencies.
>
> 1 $ = 5100 Gs., not cheap at all.
>
all that stuff is coming from hongkong and china - tons on ebay and 
auctions in all sorts of currencies with nearly free shipping, so yes it 
is cheap, does not matter where you live, you just need to look.


> Thanks.
>
> On Sun, Aug 2, 2009 at 3:07 PM, jon pounder  > wrote:
>
> Carlos Ruiz Diaz wrote:
> > Hello list,
> >
> > Why  PC modems were not used as FXO devices? Why chan_modem was
> > deprecated? it seemed a nicer option instead of buying expensive
> gateways.
>
> the digium single fxo cards and clones for about $10 ARE modems.
> you can get a sip gateway fxo + fxs in one box for about $50
>
> really - how much cheaper do you want ?
>
>
> >
> >
> >
> 
> >
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Re: [asterisk-users] Modem

[Carlos Ruiz Diaz]

I did not know that the price was that low. Anyway, for people living really
far from USA the price gets incremented twice or more and this is without
considering the conversion between currencies.

1 $ = 5100 Gs., not cheap at all.

Thanks.

On Sun, Aug 2, 2009 at 3:07 PM, jon pounder  wrote:

> Carlos Ruiz Diaz wrote:
> > Hello list,
> >
> > Why  PC modems were not used as FXO devices? Why chan_modem was
> > deprecated? it seemed a nicer option instead of buying expensive
> gateways.
>
> the digium single fxo cards and clones for about $10 ARE modems.
> you can get a sip gateway fxo + fxs in one box for about $50
>
> really - how much cheaper do you want ?
>
>
> >
> >
> > 
> >
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Re: [asterisk-users] Modem

[jon pounder]

Carlos Ruiz Diaz wrote:
> Hello list,
>
> Why  PC modems were not used as FXO devices? Why chan_modem was 
> deprecated? it seemed a nicer option instead of buying expensive gateways.

the digium single fxo cards and clones for about $10 ARE modems.
you can get a sip gateway fxo + fxs in one box for about $50

really - how much cheaper do you want ?


>
>
> 
>
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[asterisk-users] Modem

[Carlos Ruiz Diaz]

Hello list,

Why  PC modems were not used as FXO devices? Why chan_modem was deprecated?
it seemed a nicer option instead of buying expensive gateways.
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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Emrah]

Hi Tim,

I don't have any skypeforasterisk process  running. I tried to killall
-9 asterisk but it did not solve my issue.
Any other suggestions?
Thanks for your help,
Emrah
Tim Panton wrote:
> I had that too, I cured it by kill -9 'ing the skypeforasterisk
> process that was left over from
> the previous version of the beta.
>
> Hope that helps.
>
> Tim.
>
> On 2 Aug 2009, at 11:20, Emrah wrote:
>
>> I reported an issue on Mantis (#14).
>> Waiting for an update.
>> http://betareports.digium.com/mantis/view.php?id=14
>>
>> Emrah
>> Emrah wrote:
>>> Hi Thomas,
>>>
>>> I am experiencing the same problem, with the same error messages.
>>> Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686
>>>
>>> Regards,
>>> Emrah
>>> Thomas Kenyon wrote:
>>>
 Thomas Kenyon wrote:


> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
> magic number 0x25765ca0 for 0x1390e20
> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
> magic number 0x25765ca0 for 0x1390e20
>
>
>
 chef*CLI> skype show users
 Skype Users
 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad
 magic number 0x70796b73 for 0x7f4fe0044340
 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad
 magic number 0x70796b73 for 0x7f4fe0044340

 Sorry, these are the error messages.

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>>>
>>>
>>>
>>
>>
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>
> Tim Panton - Web/VoIP consultant and implementor
> www.westhawk.co.uk
>
>
>
> 
>
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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Tim Panton]

I had that too, I cured it by kill -9 'ing the skypeforasterisk  
process that was left over from

the previous version of the beta.

Hope that helps.

Tim.

On 2 Aug 2009, at 11:20, Emrah wrote:


I reported an issue on Mantis (#14).
Waiting for an update.
http://betareports.digium.com/mantis/view.php?id=14

Emrah
Emrah wrote:

Hi Thomas,

I am experiencing the same problem, with the same error messages.
Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686

Regards,
Emrah
Thomas Kenyon wrote:


Thomas Kenyon wrote:



[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20




chef*CLI> skype show users
Skype Users
[2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x70796b73 for 0x7f4fe0044340
[2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x70796b73 for 0x7f4fe0044340

Sorry, these are the error messages.

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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Pascal Bruno]

Well I think thats what the problem was, I dont have it named as eth0.  So
if your NIC is not labeled eth0 you cannot use skypeforasterisk???  Why cant
it just scan you nic handles?  Can someone point me to where I can change
the NIC name in the source file or something???



On Sun, Aug 2, 2009 at 1:05 PM, Steve Totaro  wrote:

> On Sun, Aug 2, 2009 at 12:13 PM, randulo  wrote:
>
>> On Sun, Aug 2, 2009 at 8:24 AM, Pascal Bruno wrote:
>> > So what do you think I can do to register my license? I am running
>> > Asterisk 1.6.10 on CentOS 5.
>>
>> >>> Could not generate Host-ID.
>> >>> Make sure that you have eth0 enabled.
>>
>> The MAC is used in the scheme to register and it looks like it can't
>> be read for some reason. There must be a direct channel to Digium for
>> the support of this kind, though. Have you tried contacting them?
>>
>> [waits for John Todd to chime in here...]
>>
>
>
> Is eth0 enabled?  Is it named eth0?
>
> What does ifconfig eth0 tell you?
>
> I have seen many Dell servers where the two NICs are labeled eth1 and eth2
> or whatever, but in Linux, they are backwards.  Eth2 show up as eth0 and
> eth1 shows up as eth1 in Linux.
>
> Wasted a good half hour to forty five minutes trying to figure out why I
> couldn't get the network up.
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
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Re: [asterisk-users] Converting sound files

[Christian]

Hi,
Many thanks I used it and it worked fine.
Christian


On 2009-08-02 at 11:21 Pascal Bruno wrote:

>On linux you can use Sox. Google it and resd the documentation to see  
>how you can convert files from the command line. On windows you can  
>use Switch by NCH Software. Download the trial then you can pay a  
>small fee if you want to keep it.
>
>Sent from my iPod
>
>On Aug 2, 2009, at 10:30 AM, "Christian"  wrote:
>
>> Hi all,
>> I have a set of sound files that are recorded in 16 Bit 44.1 KHz  
>> stereo and I want to convert them into 16 bit 8000 KHz mono so that  
>> i can use them in Asterisk.
>> What is the best way of doing that?
>> Many thanks,
>> Christian
>>
>>
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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Steve Totaro]

On Sun, Aug 2, 2009 at 12:13 PM, randulo  wrote:

> On Sun, Aug 2, 2009 at 8:24 AM, Pascal Bruno wrote:
> > So what do you think I can do to register my license? I am running
> > Asterisk 1.6.10 on CentOS 5.
>
> >>> Could not generate Host-ID.
> >>> Make sure that you have eth0 enabled.
>
> The MAC is used in the scheme to register and it looks like it can't
> be read for some reason. There must be a direct channel to Digium for
> the support of this kind, though. Have you tried contacting them?
>
> [waits for John Todd to chime in here...]
>


Is eth0 enabled?  Is it named eth0?

What does ifconfig eth0 tell you?

I have seen many Dell servers where the two NICs are labeled eth1 and eth2
or whatever, but in Linux, they are backwards.  Eth2 show up as eth0 and
eth1 shows up as eth1 in Linux.

Wasted a good half hour to forty five minutes trying to figure out why I
couldn't get the network up.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] how to sniff RTP and SIP traffic only

[Joe Carroll]

Wireshark will support this...

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Xavier Cardil
Sent: Monday, June 29, 2009 5:51 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to sniff RTP and SIP traffic only

Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster 
debugging ?

Thanks.
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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[randulo]

On Sun, Aug 2, 2009 at 8:24 AM, Pascal Bruno wrote:
> So what do you think I can do to register my license? I am running
> Asterisk 1.6.10 on CentOS 5.

>>> Could not generate Host-ID.
>>> Make sure that you have eth0 enabled.

The MAC is used in the scheme to register and it looks like it can't
be read for some reason. There must be a direct channel to Digium for
the support of this kind, though. Have you tried contacting them?

[waits for John Todd to chime in here...]

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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Pascal Bruno]

So what do you think I can do to register my license? I am running  
Asterisk 1.6.10 on CentOS 5.

Sent from my iPod

On Aug 2, 2009, at 3:49 AM, Thomas Kenyon   
wrote:

> Pascal Bruno wrote:
>> Unfortunately for me, I cannot register my license.  Kept saying:
>>
>> Could not generate Host-ID.
>> Make sure that you have eth0 enabled.
>>
>> Any help would be appreciated
>>
> It uses the same licensing scheme as the G.729 licenses (so as soon as
> you need to upgrade the machine, or set up LACP or VPN or any other  
> type
> of virtual interface or in the case of G.729 you change the codec to a
> newer version {since you've upgraded to a new version of asterisk that
> doesn't support older ones} that doesn't support the old name for the
> codec, you need to re-register).
>
> Or as in your case, it doesn't like the names of the network  
> interfaces.
>
> It's all a total PITA.
>
> Fwiw, the Skype channel driver stopped working on my machine a while
> ago. I never did track down the cause.
>
> When res_skypeforasterisk starts, 39 res_skypeforasterisk processes
> start and 1 skypewatcher service starts.
>
> If I start it manually after asterisk has started, usually asterisk
> segfaults, (not always).
>
> Although Sometimes it starts up properly but can't log anyone in,  
> Either
> the user is stated as Logged Out or Connection Error, usually if I  
> type
> skype show users I get the following error message:
>
> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
> magic number 0x25765ca0 for 0x1390e20
> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
> magic number 0x25765ca0 for 0x1390e20
>
> (Debian 5.0.2 x64 running kernel 2.6.30.2, asterisk 1.6.1.1 and
> skypeforasterisk-1.6.1_0.9.10-x86_64)
>
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Re: [asterisk-users] Converting sound files

[Pascal Bruno]

On linux you can use Sox. Google it and resd the documentation to see  
how you can convert files from the command line. On windows you can  
use Switch by NCH Software. Download the trial then you can pay a  
small fee if you want to keep it.

Sent from my iPod

On Aug 2, 2009, at 10:30 AM, "Christian"  wrote:

> Hi all,
> I have a set of sound files that are recorded in 16 Bit 44.1 KHz  
> stereo and I want to convert them into 16 bit 8000 KHz mono so that  
> i can use them in Asterisk.
> What is the best way of doing that?
> Many thanks,
> Christian
>
>
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Re: [asterisk-users] Converting sound files

[Doug Lytle]

Christian wrote:
> Hi all,
> I have a set of sound files that are recorded in 16 Bit 44.1 KHz stereo and I 
> want to convert them into 16 bit 8000 KHz mono so that i can use them in 
> Asterisk.
> What is the best way of doing that?
>   


http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk

Doug


-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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[asterisk-users] Converting sound files

[Christian]

Hi all,
I have a set of sound files that are recorded in 16 Bit 44.1 KHz stereo and I 
want to convert them into 16 bit 8000 KHz mono so that i can use them in 
Asterisk.
What is the best way of doing that?
Many thanks,
Christian


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Re: [asterisk-users] Different codecs for reading and writing

[Kevin P. Fleming]

Tim Panton wrote:

> The protocol expects the 2 ends to agree a single symmetrical codec
> as part of the connection setup, but it doesn't define what actually
> happens
> if the codec specified in the first (full frame) voice packet isn't what
> was agreed.

Asterisk only supports symmetric codec configuration on its internal
channels, so in Asterisk's IAX2 implementation, if a frame is received
from the other endpoint that is not in the 'expected' format a warning
is issued and the outbound direction is automatically switched to the
same format. The same is done for any protocol using RTP in Asterisk.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Different codecs for reading and writing

[Tim Panton]

On 1 Aug 2009, at 22:26, Alex Balashov wrote:


Elliot Murdock wrote:


Thank you...do you know if IAX can do this?

The reason for doing is this is to get over the adsl upload/download
discrepancy.  While G711 gives terrific quality, it is not always  
that

feasible for the upload direction, which has much more limited
bandwidth.  Accordingly, it would be possible to use G729 for upload,
but keep the higher quality codec, G711, for download.


I do not know a great deal about IAX so I will defer to the experts  
for
the definitive word on whether it is possible from the point of view  
of

its formal protocol mechanics.

However, poking around the various configuration options for IAX peers
on voip-info.org and a few other places suggests that there is no  
option

to do that with IAX, either.  It's not really something 99.9% of VoIP
users want to do.  :-)




I think you will find that it may work with Asterisk's IAX  
implementation.


The protocol expects the 2 ends to agree a single symmetrical codec
as part of the connection setup, but it doesn't define what actually  
happens
if the codec specified in the first (full frame) voice packet isn't  
what was agreed.


I have a vague memory that if the codec is one that is allowed,  
asterisk does

'the right thing' issues a warning and uses what it was given.

But, as Alex says, there is no clear way to define this in the config  
files.


You would probably do better to use Speex in both directions, but set  
the encoding quality

(in codec.conf )
parameters to be different at the 2 ends. The speex decoder should at  
the far end

should be fine with that.

see http://www.voip-info.org/wiki/view/Asterisk+config+codecs.conf

Tim.


Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk





smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Emrah]

I reported an issue on Mantis (#14).
Waiting for an update.
http://betareports.digium.com/mantis/view.php?id=14

Emrah
Emrah wrote:
> Hi Thomas,
>
> I am experiencing the same problem, with the same error messages.
> Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686
>
> Regards,
> Emrah
> Thomas Kenyon wrote:
>   
>> Thomas Kenyon wrote:
>>   
>> 
>>> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad 
>>> magic number 0x25765ca0 for 0x1390e20
>>> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad 
>>> magic number 0x25765ca0 for 0x1390e20
>>>
>>> 
>>>   
>> chef*CLI> skype show users
>> Skype Users
>> [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad 
>> magic number 0x70796b73 for 0x7f4fe0044340
>> [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad 
>> magic number 0x70796b73 for 0x7f4fe0044340
>>
>> Sorry, these are the error messages.
>>
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>>   
>> 
>
>
>   


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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Emrah]

Hi Thomas,

I am experiencing the same problem, with the same error messages.
Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686

Regards,
Emrah
Thomas Kenyon wrote:
> Thomas Kenyon wrote:
>   
>> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad 
>> magic number 0x25765ca0 for 0x1390e20
>> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad 
>> magic number 0x25765ca0 for 0x1390e20
>>
>> 
> chef*CLI> skype show users
> Skype Users
> [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad 
> magic number 0x70796b73 for 0x7f4fe0044340
> [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad 
> magic number 0x70796b73 for 0x7f4fe0044340
>
> Sorry, these are the error messages.
>
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>   


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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Thomas Kenyon]

Thomas Kenyon wrote:
> 
> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad 
> magic number 0x25765ca0 for 0x1390e20
> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad 
> magic number 0x25765ca0 for 0x1390e20
> 
chef*CLI> skype show users
Skype Users
[2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad 
magic number 0x70796b73 for 0x7f4fe0044340
[2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad 
magic number 0x70796b73 for 0x7f4fe0044340

Sorry, these are the error messages.

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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Thomas Kenyon]

Pascal Bruno wrote:
> Unfortunately for me, I cannot register my license.  Kept saying:
> 
> Could not generate Host-ID.
> Make sure that you have eth0 enabled.
> 
> Any help would be appreciated
> 
It uses the same licensing scheme as the G.729 licenses (so as soon as 
you need to upgrade the machine, or set up LACP or VPN or any other type 
of virtual interface or in the case of G.729 you change the codec to a 
newer version {since you've upgraded to a new version of asterisk that 
doesn't support older ones} that doesn't support the old name for the 
codec, you need to re-register).

Or as in your case, it doesn't like the names of the network interfaces.

It's all a total PITA.

Fwiw, the Skype channel driver stopped working on my machine a while 
ago. I never did track down the cause.

When res_skypeforasterisk starts, 39 res_skypeforasterisk processes 
start and 1 skypewatcher service starts.

If I start it manually after asterisk has started, usually asterisk 
segfaults, (not always).

Although Sometimes it starts up properly but can't log anyone in, Either 
the user is stated as Logged Out or Connection Error, usually if I type 
skype show users I get the following error message:

[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad 
magic number 0x25765ca0 for 0x1390e20
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad 
magic number 0x25765ca0 for 0x1390e20

(Debian 5.0.2 x64 running kernel 2.6.30.2, asterisk 1.6.1.1 and 
skypeforasterisk-1.6.1_0.9.10-x86_64)

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