Re: [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2

[Richard Kenner]

> Is asterisk replying differently when firmware 3.2 is used ?

No, but the phone cares with 3.2 and not with 2.6.

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Cary Fitch]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, May 04, 2011 11:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

 

 

2011/5/5 Flavio Goncalves 


but stuffing Asterisk with
many  new features on each version does not seem to be contributing to
the stability of the code or the migration to newer versions.


yes but it seems to me that code stability is improving.
Maybe next 1.10.0 version will be "production-ready" from day 1 ?
 


Flavio E. Goncalves
www.asteriskguide.com



 

Compare to which version of Windows. Patches are a never ending process

 

Cary Fitch

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Olivier]

2011/5/5 Flavio Goncalves 


> but stuffing Asterisk with
> many  new features on each version does not seem to be contributing to
> the stability of the code or the migration to newer versions.
>

yes but it seems to me that code stability is improving.
Maybe next 1.10.0 version will be "production-ready" from day 1 ?


>
> Flavio E. Goncalves
> www.asteriskguide.com
>
>
>
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Re: [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2

[Olivier]

2011/5/5 Richard Kenner 

> I recently tried to update my Aastra 57i to version 3.2 and ran into
> a problem.  It won't properly register and says "contact mismatch".
> I added "sip contact matching: 2" to aastra.cfg, but that didn't help.
>
> When I look at the SIP trace, but I see is the Aastra sending a
> REGISTER and Asterisk replying with the 401.  The phone then sends
> the REGISTER again, this time with the hash.  Asterisk now replies OK,
> but sends an OPTION packet FIRST and I think that confuses the Aastra.
>
Hi,

Is asterisk replying differently when firmware 3.2 is used ?

>
> Has anybody seen this?  Is there any way to have the packets sent in the
> proper order?
>
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[asterisk-users] Issue with Asterisk & Aastra 57i at v3.2

[Richard Kenner]

I recently tried to update my Aastra 57i to version 3.2 and ran into
a problem.  It won't properly register and says "contact mismatch".
I added "sip contact matching: 2" to aastra.cfg, but that didn't help.

When I look at the SIP trace, but I see is the Aastra sending a 
REGISTER and Asterisk replying with the 401.  The phone then sends
the REGISTER again, this time with the hash.  Asterisk now replies OK,
but sends an OPTION packet FIRST and I think that confuses the Aastra.

Has anybody seen this?  Is there any way to have the packets sent in the
proper order?

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Flavio Goncalves]

My 2 cents. All these problems seem to be lack of focus. Digium,
please stop doing everything to everyone. Too many versions, too many
features, too many code, too many bugs. Following the Pareto's
principle, 80% of the users use only 20% of the code. My suggestion is
to start thinking of Asterisk as a platform taking care of only 20% of
the code. Digium is in position to create a market place for free and
commercial Asterisk applications, drivers and modules. Look at some
other open source communities such as Joomla at
http://extensions.joomla.org, There are more than a thousand modules
maintained by the community. Imagine, do you want a multitenant
parking module? Great there is one in Digium App Store for a few
dollars. Digium could have its own commercial modules. Support for 3rd
party applications would be up to the 3rd party developers. Why iPhone
developers make money and Asterisk developer's usually don't? If
people pay for silly games in iPhones wouldn't they pay for a Unistim
driver if they have hundreds of compatible phones?

I would like to say that I have a deep respect for Asterisk and Digium
that redefined the global telephony market, but stuffing Asterisk with
many  new features on each version does not seem to be contributing to
the stability of the code or the migration to newer versions.

Flavio E. Goncalves
www.asteriskguide.com





2011/5/4 Matt Riddell :
> On 3/05/11 4:01 AM, Hans Witvliet wrote:
>>
>> Just a thought
>> If "Digium" / "the community" realy want an objective way of deciding
>> whether can/should migrate to any other version, you realy need a
>> feature-matrix (pethaps starting from version 1.2.*)
>>
>> And for every and each version a statement if it is:
>> - discontinued
>> - tested
>> - test finalized, result indicating it is fully and identically
>> functional
>> - test finalized, result indicating that this feature is changed in
>> either behaviour of configuration
>> - not yet tested.
>
> +1 From me - this would be fantastic!
>
> --
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>
> Matt Riddell
> ___
>
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>
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Re: [asterisk-users] receive faxes

[Steve Underwood]

On 05/05/2011 01:07 AM, Tzafrir Cohen wrote:

Un-top-posting,

On Wed, May 04, 2011 at 10:01:37AM -0400, vip killa wrote:

On Wed, May 4, 2011 at 9:52 AM, Danny Nicholas  wrote:

*You are “Running before you learn to walk”!  You can’t make T.38 work
(that’s ok, most other folks can’t either) but you want a free faxing
solution that does multiple channels.  Get the Free license and make that
work, then pay Digium the $10 (or whatever it is) for the ports you think
you need once the darn thing works.*

screw that i just got hylafax to work with IAXMODEM...i refuse to pay
digium a dime... supposed to be open-source right?

Asterisk's fax support has two backends. One of them is FFA mentioned
above. The other uses Steve Underwood's SpanDSP library and is
completely free (speech, beer, whatever). You don't want to pay from the
proprietary one, use the free one.

Naturally those cheap bastards at Digium wanted so badly that you buy
their FFA that they didn't bother writing the SpanDSP backend. Hmm...
well, it seems they actually did. Well, in that case they surely don't
include it in the binary packages they produce. Hmmm... they actually
do.
I've seen indications, such as at http://nerdvittles.com/?p=738 , that 
the spandsp support may not be working well these days. Can anyone 
comment on that, because all the bad stuff I've seen on this mailing 
list about FAX in 1.8 is breakage of FAX detect and the Digium FAX module?

That is not to say IAXMODEM is not a cool project on its own. Certainly
HylaFax+IAXModem is the right tool for certain scenarios, and a useful
tool generally.

Cheers,


Steve


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Re: [asterisk-users] HA Asterisk

[Michelle Dupuis]

Yes - the USB connection carries the data.  Keep in mind that the "HA" aspect 
of this product just means you can connect to two asterisk servers.  There is 
not data replication, detection of asterisk failure, etc.  (without buying more 
xorcom products).  Be sure to do your homework.  But they do make a good bank 
product.

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan 
[kaushalshri...@gmail.com]
Sent: Wednesday, May 04, 2011 9:42 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] HA Asterisk

On Mon, May 2, 2011 at 7:00 PM, Jim Dickenson 
mailto:dicken...@cfmc.com>> wrote:
On May 1, 2011, at 9:07 PM, Kaushal Shriyan wrote:

Hi Jim,

Thanks for the explanation, I have couple of questions here.

1) Does the xorcom box support 8 Port PRI E1 Interface. ?
2) Also the Primary and Secondary Asterisk Server can be any server which will 
run Asterisk or AsteriskNow (http://www.asterisk.org/asterisknow) Application 
and customizable or do i also need to buy this from Xorcom ? Not sure i 
understand that.
3) How does the xorcom box communicate with the Asterisk Server which do not 
contain any PRI Card inside the system.

Much Appreciated.

Thanks and Regards,

Kaushal


Yes Xorcom supports E1.

You can run any version of Asterisk as far as I know. I have used 1.4.x and 
ABE. The drivers are actually built in to Dahdi as supplied by Digium.

The Xorcom box communicates to both system via USB cables, one connected to 
each system.

--
Jim Dickenson
mailto:dicken...@cfmc.com


Hi Jim

Thanks for your reply. I found out from xorcom folks that we cannot re-program 
astribank, it is proprietary solution working on any given asterisk 
distribution out there. You can build linux HA cluster for asterisk then 
connect astribanks to the cluster.
They recommend twinstar (http://www.xorcom.com/optional-extras/twinstar.html) 
if we need HA solution from Xorcom.

Is there a way i can have my own customized Asterisk server behind Xorcom 
Astribank PRI box (http://www.xorcom.com/catalog/xr0111.html) and implement 
linux-ha ?

Do you have any docs for setting up linux-ha using Astribank as the PRI box and 
Customized Asterisk Server (Primary and Secondary) Setup.

Please let me know if you  need more information about my specific requirement.

Regards,

Kaushal

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Re: [asterisk-users] receive faxes

[Steve Underwood]

On 05/05/2011 03:29 AM, Lee Howard wrote:

David Backeberg wrote:

On Wed, May 4, 2011 at 12:00 PM, A J Stiles
 wrote:

(For my part, I'm actually surprised that nobody came up with a proper
protocol for encapsulating the stream of zeros and ones that make up 
a fax
transmission but rely on the precise timing inherent with a 
circuit-switched
network, into something more suitable for sending over a 
packet-switched

network.  That would have fixed it good and proper.)


They did. It's called TCP / IP.

It allows sending PDFs, and they can even be encrypted.

Faxing is for people who haven't heard of the internet.


Nobody has said that faxing couldn't use TCP/IP... and there's no 
reason why T.38 couldn't use TCP/IP.  Nobody has said that faxing 
couldn't use HTTP as a transport... or SSL... or any other kind of 
sensible mechanism.  Why in the world people try to keep faxing (data 
transfer) tied-down to audio channels by putting T.38 into H.323 or 
UDP/IP SIP beats me.
T.38 is defined to work over TCP/IP (although not TLS for some reason), 
but its rarely used. It can only really work between 2 T.38 boxes 
directly connected to the data network. To interwork with analogue FAX 
machines you need to maintain fairly tight timing, and that means 
sticking with UDP, as it does with all the other streaming stuff we do 
over UDP.


Steve


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Re: [asterisk-users] HA Asterisk

[Kaushal Shriyan]

On Mon, May 2, 2011 at 7:00 PM, Jim Dickenson  wrote:

> On May 1, 2011, at 9:07 PM, Kaushal Shriyan wrote:
>
> Hi Jim,
>
> Thanks for the explanation, I have couple of questions here.
>
> 1) Does the xorcom box support *8 Port PRI E1 Interface*. ?
> 2) Also the Primary and Secondary Asterisk Server can be any server which
> will run Asterisk or AsteriskNow (http://www.asterisk.org/asterisknow)
> Application and customizable or do i also need to buy this from Xorcom ? Not
> sure i understand that.
> 3) How does the xorcom box communicate with the Asterisk Server which do
> not contain any PRI Card inside the system.
>
> Much Appreciated.
>
> Thanks and Regards,
>
> Kaushal
>
>
> Yes Xorcom supports E1.
>
> You can run any version of Asterisk as far as I know. I have used 1.4.x and
> ABE. The drivers are actually built in to Dahdi as supplied by Digium.
>
> The Xorcom box communicates to both system via USB cables, one connected to
> each system.
>
> --
> Jim Dickenson
> mailto:dicken...@cfmc.com 
>
>
Hi Jim

Thanks for your reply. I found out from xorcom folks that we cannot
re-program astribank, it is proprietary solution working on any given
asterisk distribution out there. You can build linux HA cluster for asterisk
then connect astribanks to the cluster.
They recommend twinstar (http://www.xorcom.com/optional-extras/twinstar.html)
if we need HA solution from Xorcom.

Is there a way i can have my own customized Asterisk server behind Xorcom
Astribank PRI box (http://www.xorcom.com/catalog/xr0111.html) and implement
linux-ha ?

Do you have any docs for setting up linux-ha using Astribank as the PRI box
and Customized Asterisk Server (Primary and Secondary) Setup.

Please let me know if you  need more information about my specific
requirement.

Regards,

Kaushal
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Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

[Matt Riddell]

On 5/05/11 11:40 AM, Sherwood McGowan wrote:

ChanIsAvail + dialplan routing to call parking lot


Problem is, I think he's talking about mid call - so ChanIsAvail will 
have returned success - oh unless you can run it in the h exten?


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Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

[Sherwood McGowan]

ChanIsAvail + dialplan routing to call parking lot

On Wed, May 4, 2011 at 6:02 PM, Ira  wrote:

> At 03:21 PM 5/4/2011, you wrote:
>
>> Barring that, if the cordless phone becomes un-reachable is there a way to
>> automatically put the active call
>> on hold, or park it?  That's not the preferred solution, but it would work
>> great until I figure something else
>> out.
>>
>
> Not that it applies but I recently installed a Snom M3 and it seems to
> behave like you want. When I walk out of range and then back in the call is
> usually still there. I've not tested past that so it might hang up after an
> unknown timeout.
>
> Ira
>
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-- 
Sherwood McGowan
Telecommunications and VOIP Consultant
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Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

[Ira]

At 03:21 PM 5/4/2011, you wrote:
Barring that, if the cordless phone becomes un-reachable is there a 
way to automatically put the active call
on hold, or park it?  That's not the preferred solution, but it 
would work great until I figure something else

out.


Not that it applies but I recently installed a Snom M3 and it seems 
to behave like you want. When I walk out of range and then back in 
the call is usually still there. I've not tested past that so it 
might hang up after an unknown timeout.


Ira 



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Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

[Matt Riddell]

On 5/05/11 10:21 AM, Shawn L wrote:

  I have a situation where we have an asterisk box that is extending
several Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310).   Everything works great,
except when the cordless
phone walks out of range of one access point and into range of another
(cisco 1100 series APs).


What actually happens?  It shouldn't be disconnecting the call.

Do you have rtptimeout or something?

Qualify=x?

You could try disabling both these options.

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[asterisk-users] Park a call when sip phone becomes unreachable?

[Shawn L]

I have a situation where we have an asterisk box that is extending several
Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310).   Everything works great, except
when the cordless
phone walks out of range of one access point and into range of another
(cisco 1100 series APs).

I have another post to the list asking about how to speed up the handoff,
and keep the call active while
that's happing.  My question here is can a call be parked or placed on hold
if the SIP phone becomes
unreachable?  That way if a cordless phone user walks out of range and
'drops' the call, they have a
way to get it back

Thanks in advance
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[asterisk-users] Cordless VoIP Phones and Access Point hand-off?

[Shawn L]

 I have a situation where we have an asterisk box that is extending several
Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310).   Everything works great, except
when the cordless
phone walks out of range of one access point and into range of another
(cisco 1100 series APs).

I've been able to get virtually seamless roaming between access points to
work in the past with data
but have never tried it with voice before.  Is there a way to get asterisk
to keep the call active for a
certain period of time even after the phone becomes un-reachable then
re-attach it when the phone
comes back, or hang it up if it never comes back?

 For instance, keep the call active for 30 seconds after the phone becomes
un-reachable.  if it comes back
in 30 seconds, re-attach the active call.  If not, hang it up.

Barring that, if the cordless phone becomes un-reachable is there a way to
automatically put the active call
on hold, or park it?  That's not the preferred solution, but it would work
great until I figure something else
out.

Thanks in advance
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Re: [asterisk-users] receive faxes

[Matt Riddell]

On 5/05/11 3:02 AM, vip killa wrote:

Honestly Digium's Asterisk is not a quality project. Though it has lead
the way in innovative open-source VoIP, it's a flawed and chaotic
project. Hence, I refuse to pay Digium.


So why do you use it?

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Re: [asterisk-users] Res: Fading voice problem

[Matt Riddell]

Are you trunking the calls via IAX2 or something?

Are you using a jitter buffer?

Are you sure about the direction?

Do you get the same problem if you use something like sipp to create 30 
LAN calls and one Internet call?


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Matt Riddell]

On 3/05/11 4:01 AM, Hans Witvliet wrote:

Just a thought
If "Digium" / "the community" realy want an objective way of deciding
whether can/should migrate to any other version, you realy need a
feature-matrix (pethaps starting from version 1.2.*)

And for every and each version a statement if it is:
- discontinued
- tested
- test finalized, result indicating it is fully and identically
functional
- test finalized, result indicating that this feature is changed in
either behaviour of configuration
- not yet tested.


+1 From me - this would be fantastic!

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Re: [asterisk-users] Remove "name" part of SIP From header

[Andreas Sikkema]

On 5/4/11 7:10 PM, John Hablitzel wrote:
> exten => xxx,n,Set(CALLERID(name)=)

I'd either leave the name alone or do te following (haven't had the need
for removing it):

exten => xxx,n,Set(CALLERID(name)="")

-- 
Andreas Sikkema

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Re: [asterisk-users] Remove "name" part of SIP From header

[jjblitz]

On 5/4/2011 4:04 PM, Warren Selby wrote:
On Wed, May 4, 2011 at 12:10 PM, John Hablitzel > wrote:


Relatively new to Asterisk and SIP and am trying to run a proof of
concept using Asterisk to make an outbound call through an
Audiocodes gateway via SIP using Asterisk version 1.6.1.12.  The
specific requirements of the gateway in the configuration I am
trying to use specify that the Name part of the From header be
blank with the outbound number that needs to be dialed in the
number field of the From header. So I want it to look like this:
From: mailto:sip%3A1234567890@192.168.3.110>>;tag=xxx

However, even if I set the name to blank, using
Set(CALLERID(name)= ), Asterisk always seems to put the CallerID
number in the name field as well and here is what I get:
From: "1234567890" mailto:sip%3A1234567890@192.168.3.110>>;tag=xxx

I cannot figure out how to get the name field to be blank. Here is
the extensions.conf context that I think should work:
exten => xxx,1,Noop(Channel ID is ${CHANNEL})
exten => xxx,n,Noop(From is ${SIP_HEADER(From)})
exten => xxx,n,Set(CALLERID(num)=1234567890)
exten => xxx,n,Set(CALLERID(name)=)
exten => xxx,n,Noop(CallerID is ${CALLERID(all)})
exten => xxx,n(dialout),Dial(SIP/POTS1,60,o)
exten => xxx,n,Hangup

And my general and section from sip.conf
[general]
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
limitonpeers=yes
notifyringing=yes
maxexpirery=180
defaultexpirey=180

[POTS1]
type=friend
secret=xxx
context=pots_in
host=dynamic
dtmfmode=info
disallow=all
allow=ulaw
allow=alaw
canreinvite=no
qualify=yes
call-limit=4
rtptimeout=30

And here is the verbose CLI output from the above configuration.
-- Executing [xxx@inbound:1] NoOp("SIP/2001-0004", "Channel ID
is SIP/2001-0004") in new stack
-- Executing [xxx@inbound:2] NoOp("SIP/2001-0004", "From is
mailto:sip%3A2001@192.168.3.112>>;tag=1c354991377") in new stack
-- Executing [xxx@inbound:3] Set("SIP/2001-0004",
"CALLERID(num)=1234567890") in new stack
-- Executing [xxx@inbound:4] Set("SIP/2001-0004",
"CALLERID(name)=") in new stack
-- Executing [xxx@inbound:5] NoOp("SIP/2001-0004", "CallerID
is "" <1234567890>") in new stack
-- Executing [xxx@inbound:6] Dial("SIP/2001-0004",
"SIP/POTS1,60,o") in new stack
== Using SIP RTP CoS mark 5
-- Called POTS1
-- Got SIP response 484 "Address Incomplete" back from 192.168.3.121
== Everyone is busy/congested at this time (1:0/0/1)


It doesn't look like you're ever actually sending the number you want 
to dial?  You're setting a callerid(num), but where is the number you 
want to dial?  What happens if you change your dial command to this:


exten => xxx,n(dialout),Dial(SIP/${EXTEN}@POTS1,60,o)


--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com


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I tried your dial command and it fails as well.  This is a non-standard 
type of configuration on the gateway used for making outbound CAMA type 
of calls with DID wink and MF signalling.  All I have to do is an Invite 
to the system with the From header as described above and the gateway 
will pull the information it needs from the header.  I can make it work 
in one mode where it is expecting information in both parts (name and 
number), but it fails in another mode where it just wants the number.
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Re: [asterisk-users] asterisk call forwarding

[satish patel]

Oops!! missed your  ")" 

Sorry, It has been fixed 

From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 4 May 2011 20:43:16 +
Subject: Re: [asterisk-users] asterisk call forwarding









Hey!

I tried your statement but its not working but if i insert manually it works

exten => *72,10,Set(DB(CFIM/${fromext})=${toext})


at CLI 

- Executing [*72@from-sip:9] Wait("SIP/7102-0004", "1") in new stack
-- Executing [*72@from-sip:10] Set("SIP/7102-0004", 
"DB(CFIM/7102=7207)") in new stack


satish-desktop*CLI> database show CFIM
0 results found.
satish-desktop*CLI> 







> Date: Tue, 3 May 2011 16:41:30 -0700
> From: cwall...@lodgingcompany.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] asterisk call forwarding
> 
> On Tue, 3 May 2011 18:10:55 -0400
> Satish Patel  wrote:
> 
> > Thank you so much that solved my database issue. Now how asterisk
> > will forward call ?
> > 
> > Or I need to specify gotoif statment in my stdexten to check
> > database key and take action?
> 
> Yes, you need to write the dialplan to act on the key.  There is a
> sample out there somewhere (I've seen it) that uses the same CFIM
> database keys that you're setting.  Wherever you got the code to set
> those keys, you should be able to find the code for reading and acting
> on them...
> 
> 
> > On May 3, 2011, at 5:41 PM, Chad Wallace
> >  wrote:
> > 
> > > On Tue, 3 May 2011 18:45:32 +
> > > satish patel  wrote:
> > >
> > >>
> > >> I found following dialplan on net but somehow its not going to set
> > >> CFIM in asterisk database  (asterisk 1.8.3.3).  Any idea ?
> > >>
> > >> exten => *72,1,Answer
> > >> exten => *72,2,Wait(1)
> > >> exten => *72,3,BackGround(please-enter-your)
> > >> exten => *72,4,Playback(extension)
> > >> exten => *72,5,Read(fromext,then-press-pound)
> > >> exten => *72,6,Wait(1)
> > >> exten => *72,7,BackGround(ent-target-attendant)
> > >> exten => *72,8,Read(toext,then-press-pound)
> > >> exten => *72,9,Wait(1)
> > >> exten => *72,10,Set(DB(CFIM/${fromext}=${toext}))
> > > ^
> > > Change this line to this:
> > >
> > > exten => *72,10,Set(DB(CFIM/${fromext})=${toext})
> > >  ^
> > >
> > > The DB() function has to be closed before the equal sign.
> > >
> > >
> > >> exten => *72,11,Playback(call-fwd-unconditional)
> > >> exten => *72,12,Playback(for)
> > >> exten => *72,13,Playback(extension)
> > >> exten => *72,14,SayDigits(${fromext})
> > >> exten => *72,15,Playback(is-set-to)
> > >> exten => *72,16,SayDigits(${toext})
> > >> exten => *72,17,Hangup()
> 
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Re: [asterisk-users] asterisk call forwarding

[satish patel]

Hey!

I tried your statement but its not working but if i insert manually it works

exten => *72,10,Set(DB(CFIM/${fromext})=${toext})


at CLI 

- Executing [*72@from-sip:9] Wait("SIP/7102-0004", "1") in new stack
-- Executing [*72@from-sip:10] Set("SIP/7102-0004", 
"DB(CFIM/7102=7207)") in new stack


satish-desktop*CLI> database show CFIM
0 results found.
satish-desktop*CLI> 







> Date: Tue, 3 May 2011 16:41:30 -0700
> From: cwall...@lodgingcompany.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] asterisk call forwarding
> 
> On Tue, 3 May 2011 18:10:55 -0400
> Satish Patel  wrote:
> 
> > Thank you so much that solved my database issue. Now how asterisk
> > will forward call ?
> > 
> > Or I need to specify gotoif statment in my stdexten to check
> > database key and take action?
> 
> Yes, you need to write the dialplan to act on the key.  There is a
> sample out there somewhere (I've seen it) that uses the same CFIM
> database keys that you're setting.  Wherever you got the code to set
> those keys, you should be able to find the code for reading and acting
> on them...
> 
> 
> > On May 3, 2011, at 5:41 PM, Chad Wallace
> >  wrote:
> > 
> > > On Tue, 3 May 2011 18:45:32 +
> > > satish patel  wrote:
> > >
> > >>
> > >> I found following dialplan on net but somehow its not going to set
> > >> CFIM in asterisk database  (asterisk 1.8.3.3).  Any idea ?
> > >>
> > >> exten => *72,1,Answer
> > >> exten => *72,2,Wait(1)
> > >> exten => *72,3,BackGround(please-enter-your)
> > >> exten => *72,4,Playback(extension)
> > >> exten => *72,5,Read(fromext,then-press-pound)
> > >> exten => *72,6,Wait(1)
> > >> exten => *72,7,BackGround(ent-target-attendant)
> > >> exten => *72,8,Read(toext,then-press-pound)
> > >> exten => *72,9,Wait(1)
> > >> exten => *72,10,Set(DB(CFIM/${fromext}=${toext}))
> > > ^
> > > Change this line to this:
> > >
> > > exten => *72,10,Set(DB(CFIM/${fromext})=${toext})
> > >  ^
> > >
> > > The DB() function has to be closed before the equal sign.
> > >
> > >
> > >> exten => *72,11,Playback(call-fwd-unconditional)
> > >> exten => *72,12,Playback(for)
> > >> exten => *72,13,Playback(extension)
> > >> exten => *72,14,SayDigits(${fromext})
> > >> exten => *72,15,Playback(is-set-to)
> > >> exten => *72,16,SayDigits(${toext})
> > >> exten => *72,17,Hangup()
> 
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Re: [asterisk-users] Remove "name" part of SIP From header

[Warren Selby]

On Wed, May 4, 2011 at 12:10 PM, John Hablitzel wrote:

> Relatively new to Asterisk and SIP and am trying to run a proof of concept
> using Asterisk to make an outbound call through an Audiocodes gateway via
> SIP using Asterisk version 1.6.1.12.  The specific requirements of the
> gateway in the configuration I am trying to use specify that the Name part
> of the From header be blank with the outbound number that needs to be dialed
> in the number field of the From header. So I want it to look like this:
> From: ;tag=xxx
>
> However, even if I set the name to blank, using Set(CALLERID(name)= ),
> Asterisk always seems to put the CallerID number in the name field as well
> and here is what I get:
> From: "1234567890" ;tag=xxx
>
> I cannot figure out how to get the name field to be blank. Here is the
> extensions.conf context that I think should work:
> exten => xxx,1,Noop(Channel ID is ${CHANNEL})
> exten => xxx,n,Noop(From is ${SIP_HEADER(From)})
> exten => xxx,n,Set(CALLERID(num)=1234567890)
> exten => xxx,n,Set(CALLERID(name)=)
> exten => xxx,n,Noop(CallerID is ${CALLERID(all)})
> exten => xxx,n(dialout),Dial(SIP/POTS1,60,o)
> exten => xxx,n,Hangup
>
> And my general and section from sip.conf
> [general]
> allowoverlap=no
> udpbindaddr=0.0.0.0
> tcpenable=no
> tcpbindaddr=0.0.0.0
> srvlookup=yes
> disallow=all
> allow=ulaw
> allow=alaw
> limitonpeers=yes
> notifyringing=yes
> maxexpirery=180
> defaultexpirey=180
>
> [POTS1]
> type=friend
> secret=xxx
> context=pots_in
> host=dynamic
> dtmfmode=info
> disallow=all
> allow=ulaw
> allow=alaw
> canreinvite=no
> qualify=yes
> call-limit=4
> rtptimeout=30
>
> And here is the verbose CLI output from the above configuration.
> -- Executing [xxx@inbound:1] NoOp("SIP/2001-0004", "Channel ID is
> SIP/2001-0004") in new stack
> -- Executing [xxx@inbound:2] NoOp("SIP/2001-0004", "From is <
> sip:2001@192.168.3.112>;tag=1c354991377") in new stack
> -- Executing [xxx@inbound:3] Set("SIP/2001-0004",
> "CALLERID(num)=1234567890") in new stack
> -- Executing [xxx@inbound:4] Set("SIP/2001-0004", "CALLERID(name)=")
> in new stack
> -- Executing [xxx@inbound:5] NoOp("SIP/2001-0004", "CallerID is ""
> <1234567890>") in new stack
> -- Executing [xxx@inbound:6] Dial("SIP/2001-0004", "SIP/POTS1,60,o")
> in new stack
> == Using SIP RTP CoS mark 5
> -- Called POTS1
> -- Got SIP response 484 "Address Incomplete" back from 192.168.3.121
> == Everyone is busy/congested at this time (1:0/0/1)
>

It doesn't look like you're ever actually sending the number you want to
dial?  You're setting a callerid(num), but where is the number you want to
dial?  What happens if you change your dial command to this:

exten => xxx,n(dialout),Dial(SIP/${EXTEN}@POTS1,60,o)


-- 
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] receive faxes

[Benny Amorsen]

A J Stiles  writes:

> (For my part, I'm actually surprised that nobody came up with a proper 
> protocol for encapsulating the stream of zeros and ones that make up a fax 
> transmission but rely on the precise timing inherent with a circuit-switched 
> network, into something more suitable for sending over a packet-switched 
> network.  That would have fixed it good and proper.)

It is called T.37. However, asynchronous protocols like T.37 are a bad
match for the expectations people have of faxes.


/Benny


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Re: [asterisk-users] Multiple cards using same IRQ - getting IRQ errors and hissing

[Dean Hoover]

I'm going to upgrade the BIOS and update dahdi to the latest and greatest first.

I did look at the link you sent when I first started this mission.  It
was the basis of me looking at using ACPI to get the IRQs to change.

My maintenance window is tomorrow, so I'll let anyone who's interested
know what happened.

Dean

On Wed, May 4, 2011 at 2:31 AM, Johan Wilfer  wrote:
> On 2011-05-03 16:32, Dean Hoover wrote:
>>
>> I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an
>> HP ML110 G6 using Ubuntu Linux 10.04 LTS.
>>
>> I have two Digium TE121 single T1 port cards and a Digium AEX800
>> 8-port FXS card.  All PCI Express cards.
>>
>> Co-workers are hearing hissing sounds on some calls, and I am getting
>> IRQ errors when running "dahdi show status".
>>
>> I see that sharing IRQs for Digium cards isn't recommended, so I'm
>> trying to set it so each card gets its own.  From the few web sites
>> I've read so far, including Digium's FAQ site, I've added ACPI and
>> verified that the BIOS does not give me the ability to manually set
>> the IRQ.  I've even taken one of the TE121's out of the server (it
>> isn't being used anyways).  Everything I've done so far has not fixed
>> it.  All the cards (as well as USB1) all use IRQ 16.
>>
>> The other option given was to use setpci, but I am unfamiliar with
>> that command.  I did what I could to try and find the setting (based
>> on what the man page on Ubuntu's web site) where I could see the value
>> 16, but not getting anywhere.
>>
>> I know that this is more of an Asterisk forum than Digium.  If I need
>> to put in a case at Digium I will, but wanted to see if there were any
>> suggestions here before I pursued that.
>>
>> Any help would be appreciated.
>>
>> Dean Hoover
>>
>
> A month ago I had similar problems with a HP DL360g6 and a HP DL380g7
> running Debian 5 "Lenny".
> In the HP DL360g6 I had one TE121. I noticed IRQ misses and the problem was
> easily reproduced
> by running dahdi_maint to enable loopback and patlooptest while compiling
> asterisk to create some i/o.
>
> When I installed Debian 6 "Squeeze" instead the problem went away. Tested
> with both servers above.
> On this page I found some information about APIC (Advanced Programmable
> Interupt Controller)
> http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
> (quite old but informative)
>
> I haven't got the time to verify the root cause of the problem yet (I've
> planned to do this at the end of this month)
> but my theory is that it has something to do with the kernels APIC handling
> that was fixed between Debian 5 and 6.
>
> Maybe you experience something similar?
>
> /Johan
>
> --
> Johan Wilfer                 email: jo...@jttech.se
> JT Tech | Utvecklare         webb: http://jttech.se
> direkt: +46 31 380 91 01  support: +46 31 380 91 00
>
>
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Re: [asterisk-users] receive faxes

[Lee Howard]

David Backeberg wrote:

On Wed, May 4, 2011 at 12:00 PM, A J Stiles
 wrote:
  

(For my part, I'm actually surprised that nobody came up with a proper
protocol for encapsulating the stream of zeros and ones that make up a fax
transmission but rely on the precise timing inherent with a circuit-switched
network, into something more suitable for sending over a packet-switched
network.  That would have fixed it good and proper.)



They did. It's called TCP / IP.

It allows sending PDFs, and they can even be encrypted.

Faxing is for people who haven't heard of the internet.


Nobody has said that faxing couldn't use TCP/IP... and there's no reason 
why T.38 couldn't use TCP/IP.  Nobody has said that faxing couldn't use 
HTTP as a transport... or SSL... or any other kind of sensible 
mechanism.  Why in the world people try to keep faxing (data transfer) 
tied-down to audio channels by putting T.38 into H.323 or UDP/IP SIP 
beats me.


Lee.

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Re: [asterisk-users] best current version and motherboard/CPU compatibilities

[Doug Lytle]

|| dave cantera Mobile wrote:

I had several AMD athlons 64bit


My Myth server is this and I have an AMD X2 for my desktop.  Todays' 
chips from AMD are great.  It's just those K6 & K6-2s


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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Re: [asterisk-users] Password to be ecrypted?

[Olle E. Johansson]

4 maj 2011 kl. 19.44 skrev Robles Román, José Miguel:

> By the way, I like the implementation in iax.conf (auth=md5 ... 
> secret=x), it seems more flexible, and it enables the use of other hash 
> functions or other security algorithms.

The SIP protocol does not support any other hash functions today.

/O
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Re: [asterisk-users] asterisk-1.8 crash if no extension specified in Dial

[satish patel]

Issue created: https://issues.asterisk.org/view.php?id=19228

is there anybody could your please try this ??

-S

From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 4 May 2011 17:12:29 +
Subject: [asterisk-users] asterisk-1.8 crash if no extension specified in   
Dial








Hey All

;satish testing
exten => 7778,1,Verbose(System crash when no extension specified in dial)
exten => 7778,2,Dial(SIP/)




*CLI>   == Using SIP RTP CoS mark 5
-- Executing [7778@from-sip:1] Verbose("SIP/7527-0003", "System crash 
when no extension specified in dial") in new stack
System crash when no extension specified in dial
-- Executing [7778@from-sip:2] Dial("SIP/7527-0003", "SIP/") in new 
stack
Segmentation fault
root@shirley:~# 

  

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Re: [asterisk-users] Invalid use of undefined type when make dahdi

[CB]

> 
> On Wed, May 04, 2011 at 09:56:40PM +1200, CB wrote:
> > I am attempting to install Dahdi on a virtual machine running Centos
> 5.5 and
> > having various problems.
> >
> > yum install kernel-devel gcc make gcc-c++ libxml2-devel
> > Loaded plugins: fastestmirror
> > Loading mirror speeds from cached hostfile
> > * base: mirror.optus.net
> > * extras: mirror.optus.net
> > * rpmforge: fr2.rpmfind.net
> > * updates: mirror.optus.net
> > Setting up Install Process
> > Package kernel-devel-2.6.18-238.9.1.el5.x86_64 already installed and
> > latest version
> > Package gcc-4.1.2-50.el5.x86_64 already installed and latest version
> > Package 1:make-3.81-3.el5.x86_64 already installed and latest version
> > Package gcc-c++-4.1.2-50.el5.x86_64 already installed and latest
> version
> > Package libxml2-devel-2.6.26-2.1.2.8.el5_5.1.x86_64 already installed
> > and latest version
> > Package libxml2-devel-2.6.26-2.1.2.8.el5_5.1.i386 already installed
> and
> > latest version
> > Nothing to do
> >
> > [root@atlantis dahdi-linux-2.4.1.2]# make
> > make -C drivers/dahdi/firmware firmware-loaders
> > make[1]: Entering directory
> > `/usr/src/dahdi-linux-2.4.1.2/drivers/dahdi/firmware'
> > make[1]: Leaving directory
> > `/usr/src/dahdi-linux-2.4.1.2/drivers/dahdi/firmware'
> > You do not appear to have the sources for the 2.6.18-238.el5 kernel
> > installed.
> >
> > So we're running a different kernel...
> >
> > uname -r
> > 2.6.18-238.el5
> >
> > Downloaded kernel sources for 2.6.18-238.el5
> >
> > [user@atlantis ~]$ mkdir -p
> ~/rpmbuild/{BUILD,RPMS,SOURCES,SPECS,SRPMS}
> > [user@atlantis ~]$ echo '%_topdir %(echo $HOME)/rpmbuild' >
> > ~/.rpmmacros
> > [user@atlantis ~]$cd ~/rpmbuild/SPECS
> > [user@atlantis ~]$rpmbuild -bp --target=`uname -m` kernel-2.6.spec
> > 2> prep-err.log | tee prep-out.log
> >
> > [root@atlantis kernel-2.6.18]# cp
> > /home/user/rpmbuild/BUILD/kernel-2.6.18/linux-2.6.18.x86_64
> > /usr/src/kernels/2.6.18-238.el5-x86_64 -R
> >
> > [root@atlantis dahdi-linux-complete-2.4.1.2+2.4.1]# make all
> >
> > /usr/src/dahdi-linux-complete-
> 2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:8652:
> > error: invalid use of undefined type ‘struct module’
> > /usr/src/dahdi-linux-complete-
> 2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:8652:
> > error: ‘struct dahdi_chan’ has no member named ‘pulsecount’
> > etc etc
> >
> > This article (http://asteriskfaqs.org/2011/01/30/asterisk-
> users/invalid-use-of-undefined-type-struct-module.html) indicates that
> those errors are the result of not having CONFIG_MODULES set in the
> kernel config.
> >
> > cd /home/user/rpmbuild/BUILD/kernel-2.6.18/linux-2.6.18.x86_64
> > make menuconfig
> > [*] Enable loadable module support
> >
> > Legend: [*] built-in
> >
> > Any advice appreciated.
> 
> It still looks like the kernel that is running doesn't match the kernel
> that
> you prepped.
> 
> Can you "yum install kernel-devel-`uname -r`" after cleaning up the
> /usr/src/kernels/2.6.18-238.el5-x86_64 directory and then build?
> 
mv 2.6.18-238.el5-x86_64 /tmp
yum install kernel-devel-`uname -r`
make clean
make all
make install
service dahdi start
Loading DAHDI hardware modules:

No hardware timing source found in /proc/dahdi, loading dahdi_dummy
Running dahdi_cfg: [  OK  ]

Nice!

Thanks very much.


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Re: [asterisk-users] Password to be ecrypted?

[Robles Román , José Miguel]

> De: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] En nombre de
> Paul Hayes
> Enviado el: miércoles, 04 de mayo de 2011 17:55
> Para: asterisk-users@lists.digium.com
> Asunto: Re: [asterisk-users] Password to be ecrypted?
>
> On 03/05/11 09:09, Robles Román, José Miguel wrote:
> > Perhaps using one-way hash functions
> (http://en.wikipedia.org/wiki/Cryptographic_hash_function)
> like MD5 or SHA-x, even if you get the file with passwords
> and the code that checks them, it would be difficult to find
> a collision (a password that matches the hash). This is the
> way in which apache, for example, stores passwords (see htpasswd).
> >
> > In order to maintain compatibility, the configurarion could be
> >
> > [...}
> > secret_sha2 = ...
> >
> > Regards,
> > José Miguel
>
> I thought this already existed:
>
> http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret
>
> Although I have to admit, I've never tried using it.
>
> cheers,
> Paul.
>

I'm very sorry for the noise I've caused. I should have supposed that that 
wheel was already invented.

By the way, I like the implementation in iax.conf (auth=md5 ... secret=x), 
it seems more flexible, and it enables the use of other hash functions or other 
security algorithms.

Regards,

José Miguel

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Re: [asterisk-users] best current version and motherboard/CPU compatibilities

[|| dave cantera Mobile]

paul, doug,
I had several AMD athlons 64bit... no problems running centos, suse.  
they seem  solid on 1.4.xx... had a few intel celerons and P4s. they 
were good as well. guess I was Lucky back then!

thanks for supporting the list!
daveC

Paul Hayes wrote:

On 04/05/11 17:10, || dave cantera Mobile wrote:

doug,
why are you shaking!?!?... do you have a better recommendation?
daveC



AMD K6 CPU brings back some pretty bad memories from me too.


Doug Lytle wrote:

C F wrote:

model name : AMD-K6(tm) 3D processor


*shudder*

Doug





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Re: [asterisk-users] SIP bad request

[Mike]

Just a follow-up in case somebody else sees this: I upgraded the Polycom phone 
to the latest firmware, that did it.  I had been on the same version for almost 
a year without problems, so I don`t know if it`s the firmware version that was 
the issue or simply formatting the phone to factory default would have fixed it 
 .

 

Mike

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Friday, April 29, 2011 11:33 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP bad request

 

What I am looking for?  Here is a snippet, with some info obfuscated. I can see 
the bad request, but why there is such a message isn’t obvious.

 

 

 

<--- SIP read from UDP:23.23.23.23:23725 --->

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport

From: "JOHN SMITH" ;tag=as40e0c5af

To: "user" ;tag=372AEEC-62912E9F

CSeq: 102 INVITE

Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66

Contact: 

User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734

Allow-Events: talk,hold,conference

Accept-Language: fr-fr,fr;q=0.9,en;q=0.8

Content-Length: 0

 

 

<->

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

<--- SIP read from UDP:23.23.23.23:23725 --->

SIP/2.0 400 Bad Request

Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport

From: "JOHN SMITH" ;tag=as40e0c5af

To: "user" ;tag=372AEEC-62912E9F

CSeq: 102 INVITE

Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66

Contact: 

User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734

Accept-Language: fr-fr,fr;q=0.9,en;q=0.8

Content-Length: 0

 

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ??? ?
Sent: Friday, April 29, 2011 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP bad request

 

Try to look in 'sip set debug peer user'. 

On 29.04.2011 18:10, Mike wrote: 

Hi,

 

I have been getting reports phones ringing only a tiny moment and then going to 
voicemail.  CLI output shows:

 

-- SIP/user-0006fcdd is ringing

-- Got SIP response 400 "Bad Request" back from 23.23.23.23

-- SIP/user-0006fcdd is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)

 

Which does explain it.  How can I find the root cause of “bad request”? 
Call-limit is very high for this sip user, so I`m not reaching that limit for 
sure.

 

Mike

 
 
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[asterisk-users] asterisk-1.8 crash if no extension specified in Dial

[satish patel]

Hey All

;satish testing
exten => 7778,1,Verbose(System crash when no extension specified in dial)
exten => 7778,2,Dial(SIP/)




*CLI>   == Using SIP RTP CoS mark 5
-- Executing [7778@from-sip:1] Verbose("SIP/7527-0003", "System crash 
when no extension specified in dial") in new stack
System crash when no extension specified in dial
-- Executing [7778@from-sip:2] Dial("SIP/7527-0003", "SIP/") in new 
stack
Segmentation fault
root@shirley:~# 

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[asterisk-users] Remove "name" part of SIP From header

[John Hablitzel]

Relatively new to Asterisk and SIP and am trying to run a proof of 
concept using Asterisk to make an outbound call through an Audiocodes 
gateway via SIP using Asterisk version 1.6.1.12.  The specific 
requirements of the gateway in the configuration I am trying to use 
specify that the Name part of the From header be blank with the outbound 
number that needs to be dialed in the number field of the From header. 
So I want it to look like this:

From: ;tag=xxx

However, even if I set the name to blank, using Set(CALLERID(name)= ), 
Asterisk always seems to put the CallerID number in the name field as 
well and here is what I get:

From: "1234567890" ;tag=xxx

I cannot figure out how to get the name field to be blank. Here is the 
extensions.conf context that I think should work:

exten => xxx,1,Noop(Channel ID is ${CHANNEL})
exten => xxx,n,Noop(From is ${SIP_HEADER(From)})
exten => xxx,n,Set(CALLERID(num)=1234567890)
exten => xxx,n,Set(CALLERID(name)=)
exten => xxx,n,Noop(CallerID is ${CALLERID(all)})
exten => xxx,n(dialout),Dial(SIP/POTS1,60,o)
exten => xxx,n,Hangup

And my general and section from sip.conf
[general]
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
limitonpeers=yes
notifyringing=yes
maxexpirery=180
defaultexpirey=180

[POTS1]
type=friend
secret=xxx
context=pots_in
host=dynamic
dtmfmode=info
disallow=all
allow=ulaw
allow=alaw
canreinvite=no
qualify=yes
call-limit=4
rtptimeout=30

And here is the verbose CLI output from the above configuration.
-- Executing [xxx@inbound:1] NoOp("SIP/2001-0004", "Channel ID is 
SIP/2001-0004") in new stack
-- Executing [xxx@inbound:2] NoOp("SIP/2001-0004", "From is 
;tag=1c354991377") in new stack
-- Executing [xxx@inbound:3] Set("SIP/2001-0004", 
"CALLERID(num)=1234567890") in new stack
-- Executing [xxx@inbound:4] Set("SIP/2001-0004", "CALLERID(name)=") 
in new stack
-- Executing [xxx@inbound:5] NoOp("SIP/2001-0004", "CallerID is "" 
<1234567890>") in new stack
-- Executing [xxx@inbound:6] Dial("SIP/2001-0004", "SIP/POTS1,60,o") 
in new stack

== Using SIP RTP CoS mark 5
-- Called POTS1
-- Got SIP response 484 "Address Incomplete" back from 192.168.3.121
== Everyone is busy/congested at this time (1:0/0/1)

As you can see the Noop on the Caller ID shows that the name is blank, 
but Asterisk seems to default somehow to putting the number in the name 
field if it is blank when the Invite is created. I've also tried various 
combinations of setting CallerID(name) and (num) as well as some changes 
to settings in sip.conf for this channel that should effect caller id 
and cannot get it to clear. Is there a way to configure Asterisk not to 
do this?



Thanks in advance for any insight you can provide.

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Re: [asterisk-users] receive faxes

[Tzafrir Cohen]

Un-top-posting,

On Wed, May 04, 2011 at 10:01:37AM -0400, vip killa wrote:
> 
> On Wed, May 4, 2011 at 9:52 AM, Danny Nicholas  wrote:
> > *You are “Running before you learn to walk”!  You can’t make T.38 work
> > (that’s ok, most other folks can’t either) but you want a free faxing
> > solution that does multiple channels.  Get the Free license and make that
> > work, then pay Digium the $10 (or whatever it is) for the ports you think
> > you need once the darn thing works.*
>
> screw that i just got hylafax to work with IAXMODEM...i refuse to pay
> digium a dime... supposed to be open-source right?

Asterisk's fax support has two backends. One of them is FFA mentioned
above. The other uses Steve Underwood's SpanDSP library and is
completely free (speech, beer, whatever). You don't want to pay from the
proprietary one, use the free one.

Naturally those cheap bastards at Digium wanted so badly that you buy
their FFA that they didn't bother writing the SpanDSP backend. Hmm...
well, it seems they actually did. Well, in that case they surely don't
include it in the binary packages they produce. Hmmm... they actually
do.


That is not to say IAXMODEM is not a cool project on its own. Certainly
HylaFax+IAXModem is the right tool for certain scenarios, and a useful
tool generally.

Cheers,

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] Sangoma A400 background noise after a while

[M Shokuie]

Dear folks,

We have recently installed A400D card with 12 FXO modules, the serer is HP
DL180 G6, cards works fine but after a while all the calls get an awful
noise, you can not get what each side says. The noise cleares as soon as we
restart wanrouter but not asterisk (i mean asterisk restart does not solve).
We previsouly confronted this situation with PRI cards but not analogs,
wanpipe version is 3.5.18 and zaptel 1.4.12 also tested with recent DAHDI
with out any help. ifconfig doesnt show any overruns or errors. Once earlier
we had the same problem and come to the conclusion to change the mainboard
but this time i got mad as i couldnt change a 3000$ HP server that easy.

Is there a way i could get if there is any problem of interrupts, when i
check interrupts i could not see any shared interrupts for Snagoma card.

Anyhelp would be highly appreciated.
--
MSH
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Re: [asterisk-users] Invalid use of undefined type when make dahdi

[Shaun Ruffell]

On Wed, May 04, 2011 at 09:56:40PM +1200, CB wrote:
> I am attempting to install Dahdi on a virtual machine running Centos 5.5 and
> having various problems.
> 
> yum install kernel-devel gcc make gcc-c++ libxml2-devel
> Loaded plugins: fastestmirror
> Loading mirror speeds from cached hostfile
> * base: mirror.optus.net
> * extras: mirror.optus.net
> * rpmforge: fr2.rpmfind.net
> * updates: mirror.optus.net
> Setting up Install Process
> Package kernel-devel-2.6.18-238.9.1.el5.x86_64 already installed and
> latest version
> Package gcc-4.1.2-50.el5.x86_64 already installed and latest version
> Package 1:make-3.81-3.el5.x86_64 already installed and latest version
> Package gcc-c++-4.1.2-50.el5.x86_64 already installed and latest version
> Package libxml2-devel-2.6.26-2.1.2.8.el5_5.1.x86_64 already installed
> and latest version
> Package libxml2-devel-2.6.26-2.1.2.8.el5_5.1.i386 already installed and
> latest version
> Nothing to do
> 
> [root@atlantis dahdi-linux-2.4.1.2]# make
> make -C drivers/dahdi/firmware firmware-loaders
> make[1]: Entering directory
> `/usr/src/dahdi-linux-2.4.1.2/drivers/dahdi/firmware'
> make[1]: Leaving directory
> `/usr/src/dahdi-linux-2.4.1.2/drivers/dahdi/firmware'
> You do not appear to have the sources for the 2.6.18-238.el5 kernel
> installed.
> 
> So we're running a different kernel...
> 
> uname -r
> 2.6.18-238.el5
> 
> Downloaded kernel sources for 2.6.18-238.el5
> 
> [user@atlantis ~]$ mkdir -p ~/rpmbuild/{BUILD,RPMS,SOURCES,SPECS,SRPMS}
> [user@atlantis ~]$ echo '%_topdir %(echo $HOME)/rpmbuild' >
> ~/.rpmmacros
> [user@atlantis ~]$cd ~/rpmbuild/SPECS
> [user@atlantis ~]$rpmbuild -bp --target=`uname -m` kernel-2.6.spec
> 2> prep-err.log | tee prep-out.log
> 
> [root@atlantis kernel-2.6.18]# cp
> /home/user/rpmbuild/BUILD/kernel-2.6.18/linux-2.6.18.x86_64
> /usr/src/kernels/2.6.18-238.el5-x86_64 -R
> 
> [root@atlantis dahdi-linux-complete-2.4.1.2+2.4.1]# make all
> 
> /usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:8652:
> error: invalid use of undefined type ‘struct module’
> /usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:8652:
> error: ‘struct dahdi_chan’ has no member named ‘pulsecount’
> etc etc
> 
> This article 
> (http://asteriskfaqs.org/2011/01/30/asterisk-users/invalid-use-of-undefined-type-struct-module.html)
>  indicates that those errors are the result of not having CONFIG_MODULES set 
> in the kernel config. 
> 
> cd /home/user/rpmbuild/BUILD/kernel-2.6.18/linux-2.6.18.x86_64
> make menuconfig
> [*] Enable loadable module support
> 
> Legend: [*] built-in
> 
> Any advice appreciated.

It still looks like the kernel that is running doesn't match the kernel that
you prepped.

Can you "yum install kernel-devel-`uname -r`" after cleaning up the
/usr/src/kernels/2.6.18-238.el5-x86_64 directory and then build?

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Re: [asterisk-users] receive faxes

[David Backeberg]

On Wed, May 4, 2011 at 12:00 PM, A J Stiles
 wrote:
> (For my part, I'm actually surprised that nobody came up with a proper
> protocol for encapsulating the stream of zeros and ones that make up a fax
> transmission but rely on the precise timing inherent with a circuit-switched
> network, into something more suitable for sending over a packet-switched
> network.  That would have fixed it good and proper.)

They did. It's called TCP / IP.

It allows sending PDFs, and they can even be encrypted.

Faxing is for people who haven't heard of the internet.

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Re: [asterisk-users] best current version and motherboard/CPU compatibilities

[Doug Lytle]

|| dave cantera Mobile wrote:

why are you shaking!?!?


The AMD K6 and the AMD K6-2 were (At that time) cheaper then what Intel 
had to offer, I built many systems based on both.



Sorry, but that memory, along with the memories of running a BBS, just 
made me shudder.


Doug



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Re: [asterisk-users] receive faxes

[Steve Edwards]

On Wed, 4 May 2011, Andrew Latham wrote:


Faxing is considered a legal method of
doing business in many areas.  Maybe lobbing for more effective
digital signatures would help get faxing removed from our everyday
lives.



From dictionary.com:


lob1
[lob]  Show IPA
verb, lobbed, lob·bing, noun
–verb (used with object)
1.
Tennis . to hit (a ball) in a high arc to the back of the opponent's 
court.

2.
to fire (a missile, as a shell) in a high trajectory so that it drops onto 
a target.

3.
Cricket . to bowl (the ball) with a slow underhand motion.

Who do suggest we should be lobbing our fax machines at?

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Re: [asterisk-users] best current version and motherboard/CPU compatibilities

[Paul Hayes]

On 04/05/11 17:10, || dave cantera Mobile wrote:

doug,
why are you shaking!?!?... do you have a better recommendation?
daveC



AMD K6 CPU brings back some pretty bad memories from me too.


Doug Lytle wrote:

C F wrote:

model name : AMD-K6(tm) 3D processor


*shudder*

Doug





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Re: [asterisk-users] best current version and motherboard/CPU compatibilities

[|| dave cantera Mobile]

doug,
why are you shaking!?!?... do you have a better recommendation?
daveC

Doug Lytle wrote:

C F wrote:

model name  : AMD-K6(tm) 3D processor
   


*shudder*

Doug



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Re: [asterisk-users] receive faxes

[Andrew Latham]

On Wed, May 4, 2011 at 12:00 PM, A J Stiles
 wrote:
...
> (For my part, I'm actually surprised that nobody came up with a proper
> protocol for encapsulating the stream of zeros and ones that make up a fax
> transmission but rely on the precise timing inherent with a circuit-switched
> network, into something more suitable for sending over a packet-switched
> network.  That would have fixed it good and proper.)
>
> --
> AJS
>
> Answers come *after* questions.

AJS, thanks, love the humor.  Faxing is considered a legal method of
doing business in many areas.  Maybe lobbing for more effective
digital signatures would help get faxing removed from our everyday
lives.

-- 
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Re: [asterisk-users] receive faxes

[Danny Nicholas]

> (For my part, I'm actually surprised that nobody came up with a proper
> protocol for encapsulating the stream of zeros and ones that make up a fax
> transmission but rely on the precise timing inherent with a circuit-
> switched
> network, into something more suitable for sending over a packet-switched
> network.  That would have fixed it good and proper.)
> 
> --
> AJS
> 
> Answers come *after* questions.
> 
[Danny Nicholas] 
"Us Programmers" are too busy working on new and improved java, c, etc.
enhancements to do "simple" things like binary capture and delivery of a fax
stream instead of depending on 40 convoluted methods to try and accomplish
the same thing ;)



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[asterisk-users] Compiling extra modules

[Pan B. Christensen]

Hello,

I have been hired to fix a large and complicated installation using several 
Kamailio and Asterisk servers.

I found that I require some extra modules on some of the Asterisk servers. I 
was hoping to be able to compile only the modules needed and copy them to where 
they should be.

Asterisk was not compiled locally on each server! It was built elsewhere, added 
to svn and then checked out on each server. None of the directories are 
standard either. It will be a big job to replace all that.

When I copied the modules I had compiled and tried to load them, I got these 
errors:
*CLI> load res_odbc.so
[May 4 17:21:07] WARNING[24075]: loader.c:695 inspect_module: Module 
'res_odbc.so' was not compiled with the same compile-time options as this 
version of Asterisk.
[May 4 17:21:07] WARNING[24075]: loader.c:696 inspect_module: Module 
'res_odbc.so' will not be initialized as it may cause instability.
[May 4 17:21:07] WARNING[24075]: loader.c:734 load_resource: Module 
'res_odbc.so' could not be loaded.

Is it possible to see somewhere what compile-time options were used when 
Asterisk was compiled? Do modules count as options? If I compile Asterisk on 
two equal servers with the same options and one extra module on one server. Can 
I then copy this module to the other server and load it?

Thanks in advance!

With kind regards,
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Re: [asterisk-users] receive faxes

[A J Stiles]

On Wednesday 04 May 2011, vip killa wrote:
> Honestly Digium's Asterisk is not a quality project. Though it has lead the
> way in innovative open-source VoIP, it's a flawed and chaotic project.
> Hence, I refuse to pay Digium.

Don't worry.  You can always get your money refunded if it breaks -- and you 
even get to keep all the pieces.

> Digium seems to make a "bazillion" dollars 
> off of these flaws by selling commercial support/addons anyway... so that
> should be worth some bad karma points.

Other people seem to manage fine.  Have you considered that *you* might be the 
problem here?


(For my part, I'm actually surprised that nobody came up with a proper 
protocol for encapsulating the stream of zeros and ones that make up a fax 
transmission but rely on the precise timing inherent with a circuit-switched 
network, into something more suitable for sending over a packet-switched 
network.  That would have fixed it good and proper.)

-- 
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Answers come *after* questions.

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Re: [asterisk-users] Password to be ecrypted?

[Paul Hayes]

On 03/05/11 09:09, Robles Román, José Miguel wrote:

Perhaps using one-way hash functions 
(http://en.wikipedia.org/wiki/Cryptographic_hash_function) like MD5 or SHA-x, 
even if you get the file with passwords and the code that checks them, it would 
be difficult to find a collision (a password that matches the hash). This is 
the way in which apache, for example, stores passwords (see htpasswd).

In order to maintain compatibility, the configurarion could be

[...}
secret_sha2 = ...

Regards,
José Miguel


I thought this already existed:

http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret

Although I have to admit, I've never tried using it.

cheers,
Paul.

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Re: [asterisk-users] asterisk call forwarding

[satish patel]

Look like this is what we need 
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding 

What is conditional and unconditional forwarding ? 

> Date: Tue, 3 May 2011 16:41:30 -0700
> From: cwall...@lodgingcompany.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] asterisk call forwarding
> 
> On Tue, 3 May 2011 18:10:55 -0400
> Satish Patel  wrote:
> 
> > Thank you so much that solved my database issue. Now how asterisk
> > will forward call ?
> > 
> > Or I need to specify gotoif statment in my stdexten to check
> > database key and take action?
> 
> Yes, you need to write the dialplan to act on the key.  There is a
> sample out there somewhere (I've seen it) that uses the same CFIM
> database keys that you're setting.  Wherever you got the code to set
> those keys, you should be able to find the code for reading and acting
> on them...
> 
> 
> > On May 3, 2011, at 5:41 PM, Chad Wallace
> >  wrote:
> > 
> > > On Tue, 3 May 2011 18:45:32 +
> > > satish patel  wrote:
> > >
> > >>
> > >> I found following dialplan on net but somehow its not going to set
> > >> CFIM in asterisk database  (asterisk 1.8.3.3).  Any idea ?
> > >>
> > >> exten => *72,1,Answer
> > >> exten => *72,2,Wait(1)
> > >> exten => *72,3,BackGround(please-enter-your)
> > >> exten => *72,4,Playback(extension)
> > >> exten => *72,5,Read(fromext,then-press-pound)
> > >> exten => *72,6,Wait(1)
> > >> exten => *72,7,BackGround(ent-target-attendant)
> > >> exten => *72,8,Read(toext,then-press-pound)
> > >> exten => *72,9,Wait(1)
> > >> exten => *72,10,Set(DB(CFIM/${fromext}=${toext}))
> > > ^
> > > Change this line to this:
> > >
> > > exten => *72,10,Set(DB(CFIM/${fromext})=${toext})
> > >  ^
> > >
> > > The DB() function has to be closed before the equal sign.
> > >
> > >
> > >> exten => *72,11,Playback(call-fwd-unconditional)
> > >> exten => *72,12,Playback(for)
> > >> exten => *72,13,Playback(extension)
> > >> exten => *72,14,SayDigits(${fromext})
> > >> exten => *72,15,Playback(is-set-to)
> > >> exten => *72,16,SayDigits(${toext})
> > >> exten => *72,17,Hangup()
> 
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Re: [asterisk-users] receive faxes

[vip killa]

On Wed, May 4, 2011 at 11:13 AM, Danny Nicholas  wrote:

> Non-T.38 faxing works reasonably well.  I have some issues with some things
> at Digium as well, but I'm not going to bite the hand that feeds me.  I
> assume that Digium takes most of the known bugs out of what they charge
> folks for.  From what I read, if you had to make a living on T.38 faxing,
> you'd be in the hut in Kenya by Obama's brother.  As for Karma, we'll see..



Asterisk doesn't pay me and never will, they only make my job more difficult
because it's been in place here for so long... it would be nearly impossible
to get away from it. BTW, I'm using iaxmodem with hylafax...works much
better than T.38.
Obama's brother... damn that ain't right... that's worth some bad karma
points
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Re: [asterisk-users] receive faxes

[vip killa]

> On 4 May 2011, at 16:02, vip killa wrote:
> > Honestly Digium's Asterisk is not a quality project. Though it has lead
> the way in innovative open-source VoIP, it's a flawed and chaotic project.
> Hence, I refuse to pay Digium. Digium seems to make a "bazillion" dollars
> off of these flaws by selling commercial support/addons anyway... so that
> should be worth some bad karma points.
>
> Don't use Asterisk then.
>
>
I personally, I would never choose to use Asterisk
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Re: [asterisk-users] receive faxes

[Danny Nicholas]

Non-T.38 faxing works reasonably well.  I have some issues with some things
at Digium as well, but I'm not going to bite the hand that feeds me.  I
assume that Digium takes most of the known bugs out of what they charge
folks for.  From what I read, if you had to make a living on T.38 faxing,
you'd be in the hut in Kenya by Obama's brother.  As for Karma, we'll see...

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Eric Wieling
> Sent: Wednesday, May 04, 2011 10:07 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] receive faxes
> 
> 
> Non-T.38 faxing works well.  I assume there are reasons you must use T.38.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of vip killa
> Sent: Wednesday, May 04, 2011 11:02 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] receive faxes
> 
> Honestly Digium's Asterisk is not a quality project. Though it has lead
> the way in innovative open-source VoIP, it's a flawed and chaotic project.
> Hence, I refuse to pay Digium. Digium seems to make a "bazillion" dollars
> off of these flaws by selling commercial support/addons anyway... so that
> should be worth some bad karma points.
> 
> 
> On Wed, May 4, 2011 at 10:42 AM, Steve Edwards 
> wrote:
> 
> 
> On Wed, 4 May 2011, vip killa wrote:
> 
> 
> 
> screw that i just got hylafax to work with
> IAXMODEM...i refuse to pay digium a dime... supposed to be open-source
> right?
> 
> 
> 
> Great attitude. Should be worth about a bazillion bad karma
> points.
> 
> --
> Thanks in advance,
> --
> ---
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-
> 3867   PST
> Newline  Fax: +1-760-
> 731-3000 
> 
> --
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Re: [asterisk-users] receive faxes

[Steven Howes]

On 4 May 2011, at 15:01, vip killa wrote:
> screw that i just got hylafax to work with IAXMODEM...i refuse to pay 
> digium a dime... supposed to be open-source right?


There is so much wrong with that sentence, I don't know where to start.

On 4 May 2011, at 16:02, vip killa wrote:
> Honestly Digium's Asterisk is not a quality project. Though it has lead the 
> way in innovative open-source VoIP, it's a flawed and chaotic project. Hence, 
> I refuse to pay Digium. Digium seems to make a "bazillion" dollars off of 
> these flaws by selling commercial support/addons anyway... so that should be 
> worth some bad karma points.

Don't use Asterisk then.

S
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Re: [asterisk-users] receive faxes

[Eric Wieling]

Non-T.38 faxing works well.  I assume there are reasons you must use T.38.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, May 04, 2011 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] receive faxes

Honestly Digium's Asterisk is not a quality project. Though it has lead the way 
in innovative open-source VoIP, it's a flawed and chaotic project. Hence, I 
refuse to pay Digium. Digium seems to make a "bazillion" dollars off of these 
flaws by selling commercial support/addons anyway... so that should be worth 
some bad karma points.


On Wed, May 4, 2011 at 10:42 AM, Steve Edwards  
wrote:


On Wed, 4 May 2011, vip killa wrote:



screw that i just got hylafax to work with IAXMODEM...i 
refuse to pay digium a dime... supposed to be open-source right?



Great attitude. Should be worth about a bazillion bad karma points.

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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 
  PST
Newline  Fax: 
+1-760-731-3000 

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Re: [asterisk-users] receive faxes

[vip killa]

Honestly Digium's Asterisk is not a quality project. Though it has lead the
way in innovative open-source VoIP, it's a flawed and chaotic project.
Hence, I refuse to pay Digium. Digium seems to make a "bazillion" dollars
off of these flaws by selling commercial support/addons anyway... so that
should be worth some bad karma points.

On Wed, May 4, 2011 at 10:42 AM, Steve Edwards wrote:

> On Wed, 4 May 2011, vip killa wrote:
>
>  screw that i just got hylafax to work with IAXMODEM...i refuse to pay
>> digium a dime... supposed to be open-source right?
>>
>
> Great attitude. Should be worth about a bazillion bad karma points.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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Re: [asterisk-users] receive faxes

[Doug Lytle]

Steve Edwards wrote:

Should be worth about a bazillion bad karma points


+1

Doug


--

Ben Franklin quote:

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deserve neither Liberty nor Safety."


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Re: [asterisk-users] receive faxes

[Doug Lytle]

Eric Wieling wrote:

Does Hylafax support T.38?
   


Not at this time, no.  And, I have no need for it.

Doug


--

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


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Re: [asterisk-users] receive faxes

[Steve Edwards]

On Wed, 4 May 2011, vip killa wrote:

screw that i just got hylafax to work with IAXMODEM...i refuse to 
pay digium a dime... supposed to be open-source right?


Great attitude. Should be worth about a bazillion bad karma points.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] pickup question

[Eric Wieling]

Read the UPGRADE*.txt fles.

+101 was deprecated in 1.4

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A.Rymkus
Sent: Wednesday, May 04, 2011 10:38 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] pickup question

or, may be, I should use +101 priority? but how?

WBR
A.Rymkus

04.05.2011 17:42, A.Rymkus пишет:

Hi everybody!

I have asterisk 1.8.3.3 with pickup deadlock avoidance patch applied in 
production.

I'm use next dialplan construction
exten => *,1,PickUP(queue-number)
exten => *,2,PickUP()
Can anyone tell me how can I stop dialplan execution on the first 
priority if that priority's pickup was successfull?

I tried i priority, but no luck.

Does pickup app returns any value in success/fail result of execution?
If yes - how can I use it?

PS: google search didn't helped.

--
WBR
A.Rymkus


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Re: [asterisk-users] pickup question

[A.Rymkus]

or, may be, I should use +101 priority? but how?

WBR
A.Rymkus


04.05.2011 17:42, A.Rymkus пишет:

Hi everybody!

I have asterisk 1.8.3.3 with pickup deadlock avoidance patch applied 
in production.


I'm use next dialplan construction
exten => *,1,PickUP(queue-number)
exten => *,2,PickUP()
Can anyone tell me how can I stop dialplan execution on the first 
priority if that priority's pickup was successfull?


I tried *i* priority, but no luck.

Does pickup app returns any value in success/fail result of execution?
If yes - how can I use it?

PS: google search didn't helped.
--
WBR
A.Rymkus


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Re: [asterisk-users] receive faxes

[Steve Underwood]

Unless someone has broken something recently, you'll get better results 
with spandsp than you get with the Digium FAX package.


Steve

On 05/04/2011 09:21 PM, Satish Patel wrote:

Did you try digim fax ?

Also you can record you incoming fax via mxmonitor and analize it.

--
Sent from my iPhone

On May 4, 2011, at 8:50 AM, vip killa > wrote:


I've given up on trying T38 because there is no universal support for 
it... Can someone recommend another way of faxing without using T38?


On Tue, May 3, 2011 at 5:13 PM, satish patel > wrote:


 Enable debug and verbose on CLI ?

Did you enable and also at logger.conf
full => notice,warning,error,debug,verbose,dtmf,fax


Date: Tue, 3 May 2011 16:12:06 -0400

From: vipki...@gmail.com 
To: asterisk-users@lists.digium.com

Subject: Re: [asterisk-users] receive faxes

i have full log.. only thing that stands out are two warnings:
[May  3 16:10:40] WARNING[18176] app_fax.c: Error transmitting
fax. result=13: Unexpected message received.

[May  3 16:10:40] WARNING[18176] app_fax.c: Transmission failed




On Tue, May 3, 2011 at 4:05 PM, satish patel
mailto:satish...@hotmail.com>> wrote:

I'd enable full debug at logger.conf and try to find issue.

-S


Date: Tue, 3 May 2011 15:55:51 -0400

From: vipki...@gmail.com 
To: asterisk-users@lists.digium.com

Subject: Re: [asterisk-users] receive faxes

I tried with those settings and without... same error:

WARNING[18090]: app_fax.c:820 transmit: Transmission failed



On Tue, May 3, 2011 at 3:32 PM, satish patel
mailto:satish...@hotmail.com>> wrote:

did you set faxdetect=both or incoming

and faxbuffer=?

-S



Date: Tue, 3 May 2011 15:28:36 -0400

From: vipki...@gmail.com 
To: asterisk-users@lists.digium.com

Subject: Re: [asterisk-users] receive faxes


i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed
when trying to fax from a POTS line...

On Tue, May 3, 2011 at 3:27 PM, satish patel
mailto:satish...@hotmail.com>> wrote:

You need spandsp  i guess following is my dialplan is
working example

[fax]
exten =>
9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten =>

9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten => 9000,n,ReceiveFax(${FAXFILE})
exten => 9000,n,Hangup()




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Re: [asterisk-users] receive faxes

[Andrew Latham]

On Wed, May 4, 2011 at 10:12 AM, Eric Wieling  wrote:
>
> Does Hylafax support T.38?
>
> The free fax works just fine with DAHDI.  I've never tried to do T.38 with 
> that since it seems like it would be complicated and not give me much over 
> using DAHDI.
>

There is the t38modem[1] project.  But as others will mention, there a
many faxing patents and the t38 is not the same on all vendors. Watch
your back.

1. http://t38modem.sourceforge.net/


-- 
~~~ Andrew "lathama" Latham lath...@gmail.com ~~~

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Re: [asterisk-users] receive faxes

[vip killa]

Why use T.38 when you can use ulaw ?

On Wed, May 4, 2011 at 10:12 AM, Eric Wieling  wrote:

>
> Does Hylafax support T.38?
>
> The free fax works just fine with DAHDI.  I've never tried to do T.38 with
> that since it seems like it would be complicated and not give me much over
> using DAHDI.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
> Sent: Wednesday, May 04, 2011 10:02 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] receive faxes
>
> screw that i just got hylafax to work with IAXMODEM...i refuse to pay
> digium a dime... supposed to be open-source right?
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] receive faxes

[Eric Wieling]

Does Hylafax support T.38?

The free fax works just fine with DAHDI.  I've never tried to do T.38 with that 
since it seems like it would be complicated and not give me much over using 
DAHDI.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, May 04, 2011 10:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] receive faxes

screw that i just got hylafax to work with IAXMODEM...i refuse to pay 
digium a dime... supposed to be open-source right?


--
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Re: [asterisk-users] receive faxes

[vip killa]

screw that i just got hylafax to work with IAXMODEM...i refuse to pay
digium a dime... supposed to be open-source right?

On Wed, May 4, 2011 at 9:52 AM, Danny Nicholas  wrote:

>--
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, May 04, 2011 8:49 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] receive faxes
>
>
>
> meaning asterisk can receive only 1 fax at a time?
>
> On Wed, May 4, 2011 at 9:47 AM, Satish Patel 
> wrote:
>
> Single channel license is free.
>
> --
>
> Sent from my iPhone
>
>
> On May 4, 2011, at 9:44 AM, vip killa  wrote:
>
>  doesn't digium fax cost money?
>
> On Wed, May 4, 2011 at 9:21 AM, Satish Patel < 
> satish...@hotmail.com> wrote:
>
> Did you try digim fax ?
>
>
>
> Also you can record you incoming fax via mxmonitor and analize it.
>
> --
>
> Sent from my iPhone
>
>
> On May 4, 2011, at 8:50 AM, vip killa < 
> vipki...@gmail.com> wrote:
>
>  I've given up on trying T38 because there is no universal support for
> it... Can someone recommend another way of faxing without using T38?
>
> On Tue, May 3, 2011 at 5:13 PM, satish patel < 
> 
> satish...@hotmail.com> wrote:
>
>  Enable debug and verbose on CLI ?
>
> Did you enable and also at logger.conf
> full => notice,warning,error,debug,verbose,dtmf,fax
>  --
>
> Date: Tue, 3 May 2011 16:12:06 -0400
>
>
> From:  vipki...@gmail.com
> To:  
> asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] receive faxes
>
> i have full log.. only thing that stands out are two warnings:
> [May  3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.
> result=13: Unexpected message received.
>
> [May  3 16:10:40] WARNING[18176] app_fax.c: Transmission failed
>
>
>
>
>
>
>
> On Tue, May 3, 2011 at 4:05 PM, satish patel < 
> 
> satish...@hotmail.com> wrote:
>
> I'd enable full debug at logger.conf and try to find issue.
>
> -S
>  --
>
> Date: Tue, 3 May 2011 15:55:51 -0400
>
>
> From:  vipki...@gmail.com
> To:  
> asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] receive faxes
>
> I tried with those settings and without... same error:
>
> WARNING[18090]: app_fax.c:820 transmit: Transmission failed
>
>
>
>
>
> On Tue, May 3, 2011 at 3:32 PM, satish patel < 
> 
> satish...@hotmail.com> wrote:
>
> did you set faxdetect=both or incoming
>
> and faxbuffer=?
>
> -S
>  --
>
> Date: Tue, 3 May 2011 15:28:36 -0400
>
>
> From:  vipki...@gmail.com
> To:  
> asterisk-users@lists.digium.com
>
> Subject: Re: [asterisk-users] receive faxes
>
>
>
> i have spandsp and app_fax.so is loaded but i get:
> app_fax.c:820 transmit: Transmission failed
>
> when trying to fax from a POTS line...
>
> On Tue, May 3, 2011 at 3:27 PM, satish patel < 
> 
> satish...@hotmail.com> wrote:
>
> You need spandsp  i guess following is my dialplan is working example
>
> [fax]
> exten => 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
> exten => 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
> exten => 9000,n,ReceiveFax(${FAXFILE})
> exten => 9000,n,Hangup()
>
>  --
>
> Date: Tue, 3 May 2011 15:20:33 -0400
> From:  vipki...@gmail.com
> To:  
> asterisk-users@lists.digium.com
> Subject: [asterisk-users] receive faxes
>
>
>
> does anybody know a good tutorial on how to setup asterisk to receive faxes
> (no need to send them) ? i've tried using "app_fax.so" with T38 but i keep
> getting "Transmission failed"
>
> -- _ --
> Bandwidth and Colocation Provided by 
> 
> http://www.api-digital.com -- New to Asterisk? Join us for a live
> introductory webinar every Thurs: 
> 
> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE
> or update options visit:
> 
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>  *[Danny Nicholas] *
>
> *You are “Running before you learn to walk”!  You can’t make T.38 work
> (that’s ok, most other folks can’t either) but you want a free faxing
> solution that does multiple channels.  Get the Free license and make that
> work, then pay Digium the $10 (or whatever it is) for the ports you think
> you need once the darn thing works.*
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.di

Re: [asterisk-users] receive faxes

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, May 04, 2011 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] receive faxes

 

meaning asterisk can receive only 1 fax at a time?

On Wed, May 4, 2011 at 9:47 AM, Satish Patel  wrote:

Single channel license is free.  

--

Sent from my iPhone


On May 4, 2011, at 9:44 AM, vip killa  wrote:

doesn't digium fax cost money?

On Wed, May 4, 2011 at 9:21 AM, Satish Patel <
 satish...@hotmail.com> wrote:

Did you try digim fax ?  

 

Also you can record you incoming fax via mxmonitor and analize it. 

--

Sent from my iPhone


On May 4, 2011, at 8:50 AM, vip killa < 
vipki...@gmail.com> wrote:

I've given up on trying T38 because there is no universal support for it...
Can someone recommend another way of faxing without using T38?

On Tue, May 3, 2011 at 5:13 PM, satish patel <
  
satish...@hotmail.com> wrote:

 Enable debug and verbose on CLI ?

Did you enable and also at logger.conf 
full => notice,warning,error,debug,verbose,dtmf,fax


  _  


Date: Tue, 3 May 2011 16:12:06 -0400


From:    
vipki...@gmail.com
To:  
 asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes

i have full log.. only thing that stands out are two warnings:
[May  3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.
result=13: Unexpected message received.

[May  3 16:10:40] WARNING[18176] app_fax.c: Transmission failed

 

 

 

On Tue, May 3, 2011 at 4:05 PM, satish patel <
  
satish...@hotmail.com> wrote:

I'd enable full debug at logger.conf and try to find issue.

-S


  _  


Date: Tue, 3 May 2011 15:55:51 -0400


From:    
vipki...@gmail.com
To:  
 asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes

I tried with those settings and without... same error:

WARNING[18090]: app_fax.c:820 transmit: Transmission failed

 

 

On Tue, May 3, 2011 at 3:32 PM, satish patel <
  
satish...@hotmail.com> wrote:

did you set faxdetect=both or incoming 

and faxbuffer=?

-S 


  _  


Date: Tue, 3 May 2011 15:28:36 -0400


From:    
vipki...@gmail.com
To:  
 asterisk-users@lists.digium.com

Subject: Re: [asterisk-users] receive faxes



i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed

when trying to fax from a POTS line...

On Tue, May 3, 2011 at 3:27 PM, satish patel <
  
satish...@hotmail.com> wrote:

You need spandsp  i guess following is my dialplan is working example 

[fax]
exten => 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten => 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten => 9000,n,ReceiveFax(${FAXFILE})
exten => 9000,n,Hangup()




  _  


Date: Tue, 3 May 2011 15:20:33 -0400
From:    
vipki...@gmail.com
To:  
 asterisk-users@lists.digium.com
Subject: [asterisk-users] receive faxes



does anybody know a good tutorial on how to setup asterisk to receive faxes
(no need to send them) ? i've tried using "app_fax.so" with T38 but i keep
getting "Transmission failed" 

-- _ --
Bandwidth and Colocation Provided by  
 http://www.api-digital.com -- New to Asterisk?
Join us for a live introductory webinar every Thurs:
  
http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or
update options visit:


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





[Danny Nicholas] 

You are "Running before you learn to walk"!  You can't make T.38 work
(that's ok, most other folks can't either) but you want a free faxing
solution that does multiple channels.  Get the Free license and make that
work, then pay Digium the $10 (or whatever it is) for the ports you think
you need once the darn thing works.

 

--

Re: [asterisk-users] receive faxes

[vip killa]

meaning asterisk can receive only 1 fax at a time?

On Wed, May 4, 2011 at 9:47 AM, Satish Patel  wrote:

> Single channel license is free.
>
> --
> Sent from my iPhone
>
> On May 4, 2011, at 9:44 AM, vip killa  wrote:
>
> doesn't digium fax cost money?
>
> On Wed, May 4, 2011 at 9:21 AM, Satish Patel < 
> satish...@hotmail.com> wrote:
>
>> Did you try digim fax ?
>>
>> Also you can record you incoming fax via mxmonitor and analize it.
>>
>> --
>> Sent from my iPhone
>>
>> On May 4, 2011, at 8:50 AM, vip killa < 
>> vipki...@gmail.com> wrote:
>>
>> I've given up on trying T38 because there is no universal support for
>> it... Can someone recommend another way of faxing without using T38?
>>
>> On Tue, May 3, 2011 at 5:13 PM, satish patel < 
>> 
>> satish...@hotmail.com> wrote:
>>
>>>   Enable debug and verbose on CLI ?
>>>
>>> Did you enable and also at logger.conf
>>> full => notice,warning,error,debug,verbose,dtmf,fax
>>>
>>> --
>>> Date: Tue, 3 May 2011 16:12:06 -0400
>>>
>>> From:  vipki...@gmail.com
>>> To:  
>>> asterisk-users@lists.digium.com
>>> Subject: Re: [asterisk-users] receive faxes
>>>
>>> i have full log.. only thing that stands out are two warnings:
>>> [May  3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.
>>> result=13: Unexpected message received.
>>>
>>> [May  3 16:10:40] WARNING[18176] app_fax.c: Transmission failed
>>>
>>>
>>>
>>>
>>> On Tue, May 3, 2011 at 4:05 PM, satish patel < 
>>> 
>>> satish...@hotmail.com> wrote:
>>>
>>>  I'd enable full debug at logger.conf and try to find issue.
>>>
>>> -S
>>>
>>> --
>>> Date: Tue, 3 May 2011 15:55:51 -0400
>>>
>>> From:  vipki...@gmail.com
>>> To:  
>>> asterisk-users@lists.digium.com
>>> Subject: Re: [asterisk-users] receive faxes
>>>
>>> I tried with those settings and without... same error:
>>>
>>> WARNING[18090]: app_fax.c:820 transmit: Transmission failed
>>>
>>>
>>>
>>> On Tue, May 3, 2011 at 3:32 PM, satish patel < 
>>> 
>>> satish...@hotmail.com> wrote:
>>>
>>>  did you set faxdetect=both or incoming
>>>
>>> and faxbuffer=?
>>>
>>> -S
>>>
>>> --
>>> Date: Tue, 3 May 2011 15:28:36 -0400
>>>
>>> From:  vipki...@gmail.com
>>> To:  
>>> asterisk-users@lists.digium.com
>>> Subject: Re: [asterisk-users] receive faxes
>>>
>>>
>>> i have spandsp and app_fax.so is loaded but i get:
>>> app_fax.c:820 transmit: Transmission failed
>>> when trying to fax from a POTS line...
>>>
>>> On Tue, May 3, 2011 at 3:27 PM, satish patel < 
>>> 
>>> satish...@hotmail.com> wrote:
>>>
>>>  You need spandsp  i guess following is my dialplan is working example
>>>
>>> [fax]
>>> exten => 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
>>> exten =>
>>> 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
>>> exten => 9000,n,ReceiveFax(${FAXFILE})
>>> exten => 9000,n,Hangup()
>>>
>>>
>>> --
>>> Date: Tue, 3 May 2011 15:20:33 -0400
>>> From:  vipki...@gmail.com
>>> To:  
>>> asterisk-users@lists.digium.com
>>> Subject: [asterisk-users] receive faxes
>>>
>>>
>>> does anybody know a good tutorial on how to setup asterisk to receive
>>> faxes (no need to send them) ? i've tried using "app_fax.so" with T38 but i
>>> keep getting "Transmission failed"
>>> -- _
>>> -- Bandwidth and Colocation Provided by 
>>> 
>>> http://www.api-digital.com -- New to Asterisk? Join us for a live
>>> introductory webinar every Thurs: 
>>> 
>>> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE
>>> or update options visit:
>>> 
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by 
>>> 
>>> http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>   
>>> http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   
>>> 
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>
>>> -- _
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>>> 
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Re: [asterisk-users] receive faxes

[Satish Patel]

Single channel license is free.

--
Sent from my iPhone

On May 4, 2011, at 9:44 AM, vip killa  wrote:


doesn't digium fax cost money?

On Wed, May 4, 2011 at 9:21 AM, Satish Patel   
wrote:

Did you try digim fax ?

Also you can record you incoming fax via mxmonitor and analize it.

--
Sent from my iPhone

On May 4, 2011, at 8:50 AM, vip killa  wrote:

I've given up on trying T38 because there is no universal support  
for it... Can someone recommend another way of faxing without using  
T38?


On Tue, May 3, 2011 at 5:13 PM, satish patel  
 wrote:

 Enable debug and verbose on CLI ?

Did you enable and also at logger.conf
full => notice,warning,error,debug,verbose,dtmf,fax

Date: Tue, 3 May 2011 16:12:06 -0400

From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes

i have full log.. only thing that stands out are two warnings:
[May  3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.  
result=13: Unexpected message received.


[May  3 16:10:40] WARNING[18176] app_fax.c: Transmission failed




On Tue, May 3, 2011 at 4:05 PM, satish patel  
 wrote:

I'd enable full debug at logger.conf and try to find issue.

-S

Date: Tue, 3 May 2011 15:55:51 -0400

From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes

I tried with those settings and without... same error:

WARNING[18090]: app_fax.c:820 transmit: Transmission failed



On Tue, May 3, 2011 at 3:32 PM, satish patel  
 wrote:

did you set faxdetect=both or incoming

and faxbuffer=?

-S

Date: Tue, 3 May 2011 15:28:36 -0400

From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes


i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed
when trying to fax from a POTS line...

On Tue, May 3, 2011 at 3:27 PM, satish patel  
 wrote:

You need spandsp  i guess following is my dialplan is working example

[fax]
exten => 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten => 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID 
(num)})

exten => 9000,n,ReceiveFax(${FAXFILE})
exten => 9000,n,Hangup()


Date: Tue, 3 May 2011 15:20:33 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] receive faxes


does anybody know a good tutorial on how to setup asterisk to  
receive faxes (no need to send them) ? i've tried using  
"app_fax.so" with T38 but i keep getting "Transmission failed"
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Re: [asterisk-users] receive faxes

[Danny Nicholas]

Digium has a 1-port Free Fax for Asterisk - FFA

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, May 04, 2011 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] receive faxes

 

doesn't digium fax cost money?

On Wed, May 4, 2011 at 9:21 AM, Satish Patel  wrote:

Did you try digim fax ?  

 

Also you can record you incoming fax via mxmonitor and analize it. 

--

Sent from my iPhone


On May 4, 2011, at 8:50 AM, vip killa  wrote:

I've given up on trying T38 because there is no universal support for it...
Can someone recommend another way of faxing without using T38?

On Tue, May 3, 2011 at 5:13 PM, satish patel <
 satish...@hotmail.com> wrote:

 Enable debug and verbose on CLI ?

Did you enable and also at logger.conf 
full => notice,warning,error,debug,verbose,dtmf,fax


  _  


Date: Tue, 3 May 2011 16:12:06 -0400


From:   vipki...@gmail.com
To:  
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes

i have full log.. only thing that stands out are two warnings:
[May  3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.
result=13: Unexpected message received.

[May  3 16:10:40] WARNING[18176] app_fax.c: Transmission failed

 

 

 

On Tue, May 3, 2011 at 4:05 PM, satish patel <
 satish...@hotmail.com> wrote:

I'd enable full debug at logger.conf and try to find issue.

-S


  _  


Date: Tue, 3 May 2011 15:55:51 -0400


From:   vipki...@gmail.com
To:  
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes

I tried with those settings and without... same error:

WARNING[18090]: app_fax.c:820 transmit: Transmission failed

 

 

On Tue, May 3, 2011 at 3:32 PM, satish patel <
 satish...@hotmail.com> wrote:

did you set faxdetect=both or incoming 

and faxbuffer=?

-S 


  _  


Date: Tue, 3 May 2011 15:28:36 -0400


From:   vipki...@gmail.com
To:  
asterisk-users@lists.digium.com

Subject: Re: [asterisk-users] receive faxes



i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed

when trying to fax from a POTS line...

On Tue, May 3, 2011 at 3:27 PM, satish patel <
 satish...@hotmail.com> wrote:

You need spandsp  i guess following is my dialplan is working example 

[fax]
exten => 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten => 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten => 9000,n,ReceiveFax(${FAXFILE})
exten => 9000,n,Hangup()




  _  


Date: Tue, 3 May 2011 15:20:33 -0400
From:   vipki...@gmail.com
To:  
asterisk-users@lists.digium.com
Subject: [asterisk-users] receive faxes



does anybody know a good tutorial on how to setup asterisk to receive faxes
(no need to send them) ? i've tried using "app_fax.so" with T38 but i keep
getting "Transmission failed" 

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Re: [asterisk-users] receive faxes

[vip killa]

doesn't digium fax cost money?

On Wed, May 4, 2011 at 9:21 AM, Satish Patel  wrote:

> Did you try digim fax ?
>
> Also you can record you incoming fax via mxmonitor and analize it.
>
> --
> Sent from my iPhone
>
> On May 4, 2011, at 8:50 AM, vip killa  wrote:
>
> I've given up on trying T38 because there is no universal support for it...
> Can someone recommend another way of faxing without using T38?
>
> On Tue, May 3, 2011 at 5:13 PM, satish patel < 
> satish...@hotmail.com> wrote:
>
>>   Enable debug and verbose on CLI ?
>>
>> Did you enable and also at logger.conf
>> full => notice,warning,error,debug,verbose,dtmf,fax
>>
>> --
>> Date: Tue, 3 May 2011 16:12:06 -0400
>>
>> From: vipki...@gmail.com
>> To: asterisk-users@lists.digium.com
>> Subject: Re: [asterisk-users] receive faxes
>>
>> i have full log.. only thing that stands out are two warnings:
>> [May  3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.
>> result=13: Unexpected message received.
>>
>> [May  3 16:10:40] WARNING[18176] app_fax.c: Transmission failed
>>
>>
>>
>>
>> On Tue, May 3, 2011 at 4:05 PM, satish patel < 
>> satish...@hotmail.com> wrote:
>>
>>  I'd enable full debug at logger.conf and try to find issue.
>>
>> -S
>>
>> --
>> Date: Tue, 3 May 2011 15:55:51 -0400
>>
>> From: vipki...@gmail.com
>> To: asterisk-users@lists.digium.com
>> Subject: Re: [asterisk-users] receive faxes
>>
>> I tried with those settings and without... same error:
>>
>> WARNING[18090]: app_fax.c:820 transmit: Transmission failed
>>
>>
>>
>> On Tue, May 3, 2011 at 3:32 PM, satish patel < 
>> satish...@hotmail.com> wrote:
>>
>>  did you set faxdetect=both or incoming
>>
>> and faxbuffer=?
>>
>> -S
>>
>> --
>> Date: Tue, 3 May 2011 15:28:36 -0400
>>
>> From: vipki...@gmail.com
>> To: asterisk-users@lists.digium.com
>> Subject: Re: [asterisk-users] receive faxes
>>
>>
>> i have spandsp and app_fax.so is loaded but i get:
>> app_fax.c:820 transmit: Transmission failed
>> when trying to fax from a POTS line...
>>
>> On Tue, May 3, 2011 at 3:27 PM, satish patel < 
>> satish...@hotmail.com> wrote:
>>
>>  You need spandsp  i guess following is my dialplan is working example
>>
>> [fax]
>> exten => 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
>> exten => 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
>> exten => 9000,n,ReceiveFax(${FAXFILE})
>> exten => 9000,n,Hangup()
>>
>>
>> --
>> Date: Tue, 3 May 2011 15:20:33 -0400
>> From: vipki...@gmail.com
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] receive faxes
>>
>>
>> does anybody know a good tutorial on how to setup asterisk to receive
>> faxes (no need to send them) ? i've tried using "app_fax.so" with T38 but i
>> keep getting "Transmission failed"
>> -- _
>> -- Bandwidth and Colocation Provided by 
>> http://www.api-digital.com -- New to Asterisk? Join us for a live
>> introductory webinar every Thurs: 
>> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE
>> or update options visit:
>> 
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by 
>> http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
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>> introductory webinar every Thurs: 
>> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE
>> or update options visit:
>> 
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by 
>> http://www.api-digital.com --
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>>   
>> http://www.asterisk.org/hello
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>> To UNSUBSCRIBE or update options visit:
>>   
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> -- __

[asterisk-users] pickup question

[A.Rymkus]

Hi everybody!

I have asterisk 1.8.3.3 with pickup deadlock avoidance patch applied in 
production.


I'm use next dialplan construction
exten => *,1,PickUP(queue-number)
exten => *,2,PickUP()
Can anyone tell me how can I stop dialplan execution on the first 
priority if that priority's pickup was successfull?


I tried *i* priority, but no luck.

Does pickup app returns any value in success/fail result of execution?
If yes - how can I use it?

PS: google search didn't helped.

--
WBR
A.Rymkus

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Re: [asterisk-users] receive faxes

[Satish Patel]

Did you try digim fax ?

Also you can record you incoming fax via mxmonitor and analize it.

--
Sent from my iPhone

On May 4, 2011, at 8:50 AM, vip killa  wrote:

I've given up on trying T38 because there is no universal support  
for it... Can someone recommend another way of faxing without using  
T38?


On Tue, May 3, 2011 at 5:13 PM, satish patel   
wrote:

 Enable debug and verbose on CLI ?

Did you enable and also at logger.conf
full => notice,warning,error,debug,verbose,dtmf,fax

Date: Tue, 3 May 2011 16:12:06 -0400

From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes

i have full log.. only thing that stands out are two warnings:
[May  3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.  
result=13: Unexpected message received.


[May  3 16:10:40] WARNING[18176] app_fax.c: Transmission failed




On Tue, May 3, 2011 at 4:05 PM, satish patel   
wrote:

I'd enable full debug at logger.conf and try to find issue.

-S

Date: Tue, 3 May 2011 15:55:51 -0400

From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes

I tried with those settings and without... same error:

WARNING[18090]: app_fax.c:820 transmit: Transmission failed



On Tue, May 3, 2011 at 3:32 PM, satish patel   
wrote:

did you set faxdetect=both or incoming

and faxbuffer=?

-S

Date: Tue, 3 May 2011 15:28:36 -0400

From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes


i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed
when trying to fax from a POTS line...

On Tue, May 3, 2011 at 3:27 PM, satish patel   
wrote:

You need spandsp  i guess following is my dialplan is working example

[fax]
exten => 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten => 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID 
(num)})

exten => 9000,n,ReceiveFax(${FAXFILE})
exten => 9000,n,Hangup()


Date: Tue, 3 May 2011 15:20:33 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] receive faxes


does anybody know a good tutorial on how to setup asterisk to  
receive faxes (no need to send them) ? i've tried using "app_fax.so"  
with T38 but i keep getting "Transmission failed"
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Re: [asterisk-users] How to debug MixMonitor misbehaviour

[Bruce B]

Thanks for the input. I think that works as my other recordings work. I will
test that again regardless.

Is there no real other way to know why MixMonitor fails or look more into
it?

Regards,
Bruce

On Wed, May 4, 2011 at 5:03 AM, salaheddine elharit <
salah.elharit...@gmail.com> wrote:

> hi
>
> you can add this in extenssion.conf
>
>
> exten => 223,1,Answer()
>
> exten => 223,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
>
> exten => 223,3,Dial(SIP/223)
>
> exten => 223,4,Hangup()
>
> i can record without any issue in /var/spool/asterisk/monitor
>
>
> 2011/5/4 Bruce B 
>
>> Thanks for the input.
>>
>> Yes, I did call out many times, but the recording doesn't happen even
>> after the call is bridged and there is two way audio. I also took out the
>> "b" option and so it should recording the ringing right (even before call is
>> bridged) but it doesn't do that or any recording at all.
>>
>> Any other suggestions as to what I can do to see why this is not
>> recording?
>>
>> Regards,
>>
>>
>> On Tue, May 3, 2011 at 2:13 AM, virendra bhati wrote:
>>
>>> Hi,
>>>
>>> As per your Dialplan MixMonitor will work after call bridge, In you case
>>> still call is not bridge. That's why MixMonitor is waiting of call bridge...
>>>
>>> *
>>> MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)
>>> option b=>** A bridge flag allows recording to only take place when the
>>> channel is bridged.*
>>>
>>> So just for test make sip call and start mixmonitor to test the recorded
>>> file.
>>> default path od recording id
>>> *
>>>  /var/spool/asterisk/monitor/
>>>
>>> *
>>>  On Tue, May 3, 2011 at 10:40 AM, Bruce B  wrote:
>>>
  Hi everyone,

 For some reason MixMonitor doesn't record when it should; It actually
 shows the MixMonitor line just fine on the CLI. How can MixMonitor be
 debugged for things like privilege issues or filename issues?

 **I had this working at one point and then stopped working. Not sure
 what I changed.

 System Info:
 Asterisk 1.4.21.2
 Queuemetrics 1.6.3.0


 [queuedial]
  ; this piece of dialplan is just a calling hook into the
 [qm-queuedial] context that actually does the
 ; outbound dialing - replace as needed - just fill in the same
 variables.
 exten => _XXX.,1,Set(QDIALER_QUEUE=q-${EXTEN:0:3})
 exten => _XXX.,n,Set(QDIALER_NUMBER=${EXTEN:3})
 exten => _XXX.,n,Set(QDIALER_AGENT=Agent/${CALLERID(num)})
 exten => _XXX.,n,Set(QDIALER_CHANNEL=ZAP/g0/${QDIALER_NUMBER})
 exten => _XXX.,n,Set(QueueName=${QDIALER_QUEUE})
 *exten => _XXX.,n,MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)*
 exten => _XXX.,n,Goto(qm-queuedial,s,1)

 CLI output:
  -- Called 4904166356574@queuedial/n
 -- Executing [4904166356574@queuedial:1]
 Set("Local/4904166356574@queuedial-d851,2", "QDIALER_QUEUE=q-490") in
 new stack
 -- Executing [4904166356574@queuedial:2]
 Set("Local/4904166356574@queuedial-d851,2",
 "QDIALER_NUMBER=4166356574") in new stack
 -- Executing [4904166356574@queuedial:3]
 Set("Local/4904166356574@queuedial-d851,2",
 "QDIALER_AGENT=Agent/19053640558") in new stack
 -- Executing [4904166356574@queuedial:4]
 Set("Local/4904166356574@queuedial-d851,2",
 "QDIALER_CHANNEL=ZAP/g0/4166356574") in new stack
 -- Executing [4904166356574@queuedial:5]
 Set("Local/4904166356574@queuedial-d851,2", "QueueName=q-490") in new
 stack
 *-- Executing [4904166356574@queuedial:6]
 MixMonitor("Local/4904166356574@queuedial-d851,2",
 "Q-q-490-1304399098.18.WAV|b|") in new stack*
 -- Executing [4904166356574@queuedial:7]
 Goto("Local/4904166356574@queuedial-d851,2", "qm-queuedial|s|1") in new
 stack
 -- Goto (qm-queuedial,s,1)

 Trying to locate file:
  root@pbx:~ $ updatedb
 root@pbx:~ $ locate Q-q-490-1304399098.18.WAV
 root@pbx:~ $ ls /var/spool/asterisk/monitor/Q-q*
 ls: /var/spool/asterisk/monitor/Q-q*: No such file or directory

 I also turned on the Debug but I couldn't see anything out of the norm.
 As you can see above the CLI output is just fine.

 Thanks,
 Bruce

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>>>
>>>
>>>
>>> --
>>>
>>>
>>>
>>> -
>>> Thanks and regards
>>>
>>>  Virendra Bhati
>>> +91-9172341457
>>>
>>>
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Re: [asterisk-users] receive faxes

[vip killa]

I've given up on trying T38 because there is no universal support for it...
Can someone recommend another way of faxing without using T38?

On Tue, May 3, 2011 at 5:13 PM, satish patel  wrote:

>   Enable debug and verbose on CLI ?
>
> Did you enable and also at logger.conf
> full => notice,warning,error,debug,verbose,dtmf,fax
>
> --
> Date: Tue, 3 May 2011 16:12:06 -0400
>
> From: vipki...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] receive faxes
>
> i have full log.. only thing that stands out are two warnings:
> [May  3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.
> result=13: Unexpected message received.
>
> [May  3 16:10:40] WARNING[18176] app_fax.c: Transmission failed
>
>
>
>
> On Tue, May 3, 2011 at 4:05 PM, satish patel wrote:
>
>  I'd enable full debug at logger.conf and try to find issue.
>
> -S
>
> --
> Date: Tue, 3 May 2011 15:55:51 -0400
>
> From: vipki...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] receive faxes
>
> I tried with those settings and without... same error:
>
> WARNING[18090]: app_fax.c:820 transmit: Transmission failed
>
>
>
> On Tue, May 3, 2011 at 3:32 PM, satish patel wrote:
>
>  did you set faxdetect=both or incoming
>
> and faxbuffer=?
>
> -S
>
> --
> Date: Tue, 3 May 2011 15:28:36 -0400
>
> From: vipki...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] receive faxes
>
>
> i have spandsp and app_fax.so is loaded but i get:
> app_fax.c:820 transmit: Transmission failed
> when trying to fax from a POTS line...
>
> On Tue, May 3, 2011 at 3:27 PM, satish patel wrote:
>
>  You need spandsp  i guess following is my dialplan is working example
>
> [fax]
> exten => 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
> exten => 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
> exten => 9000,n,ReceiveFax(${FAXFILE})
> exten => 9000,n,Hangup()
>
>
> --
> Date: Tue, 3 May 2011 15:20:33 -0400
> From: vipki...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] receive faxes
>
>
> does anybody know a good tutorial on how to setup asterisk to receive faxes
> (no need to send them) ? i've tried using "app_fax.so" with T38 but i keep
> getting "Transmission failed"
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Re: [asterisk-users] asterisk 1.4.35 to 1.4.41

[Eric Wieling]

Here is the bug report

https://issues.asterisk.org/view.php?id=19171


Please add a comment to the bug indicating that you are also experiencing the 
issue with asterisk 1.4.35 to 1.4.41

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel
Sent: Wednesday, May 04, 2011 8:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk 1.4.35 to 1.4.41

Look like codec mismatch issue.

--
Sent from my iPhone

On May 3, 2011, at 9:55 PM, Jerry Geis  wrote:

> Under 1.4.35 I get this message printed MANY times
> [May  3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
> type 4, while native formats is 0x1000 (g722)(4096) read/write =
> 0x1000 (g722)(4096)/0x1000 (g722)(4096)
> [May  3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
> type 4, while native formats is 0x1000 (g722)(4096) read/write =
> 0x1000 (g722)(4096)/0x1000 (g722)(4096)
> [May  3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
> type 4, while native formats is 0x1000 (g722)(4096) read/write =
> 0x1000 (g722)(4096)/0x1000 (g722)(4096)
> [May  3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
> type 4, while native formats is 0x1000 (g722)(4096) read/write =
> 0x1000 (g722)(4096)/0x1000 (g722)(4096)
> [May  3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
> type 4, while native formats is 0x1000 (g722)(4096) read/write =
> 0x1000 (g722)(4096)/0x1000 (g722)(4096)
>
> Under 1.4.41 I get an error and hang up doing the exact same thing.
>
> All I am doing Is calling a cell phone over the PRI then dialing my
> SIP/524 extension.
>
>
> This is from 1.4.35
>  > Channel DAHDI/18-1 was answered.
>   -- Executing [smvoice_callprogress@smvoice-dialout:1] GotoIf
> ("DAHDI/18-1", "1?smvoice_callprogress|3:smvoice_callprogress|2") in
> new stack
>   -- Goto (smvoice-dialout,smvoice_callprogress,3)
>   -- Executing [smvoice_callprogress@smvoice-dialout:3] AGI("DAHDI/
> 18-1", "smvoice) in new stack
>   -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
>   -- Playing '/home/silentm/record/please_press/
> one_to_call.' (escape_digits=0123456789*#) (sample_offset 0)
> [May  3 21:47:38] DTMF[21746]: channel.c:2368 __ast_read: DTMF end
> '1' received on DAHDI/18-1, duration 0 ms
> [May  3 21:47:38] DTMF[21746]: channel.c:2423 __ast_read: DTMF end
> accepted without begin '1' on DAHDI/18-1
> [May  3 21:47:38] DTMF[21746]: channel.c:2434 __ast_read: DTMF end
> passthrough '1' on DAHDI/18-1
>   -- Playing '/tmp/smvoice.21747_0' (escape_digits=0123456789#)
> (sample_offset 0)
> [May  3 21:47:41] ERROR[21746]: utils.c:968 ast_carefulwrite: write
> () returned error: Broken pipe
>   -- AGI Script smvoice completed, returning 0
>   -- Executing [smvoice_dial_goto_voicemail@smvoice-dialout:1] Dial
> ("DAHDI/18-1", "SIP/524|30|tT") in new stack
>   -- Called 524
> [May  3 21:47:41] WARNING[21746]: channel.c:3782
> ast_channel_make_compatible: No path to translate from SIP/
> 524-0001(4096) to DAHDI/18-1(4)
> [May  3 21:47:41] WARNING[21746]: chan_sip.c:3890 sip_write: Asked
> to transmit frame type 4, while native formats is 0x1000 (g722)
> (4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096)
> [May  3 21:47:41] WARNING[21746]: chan_sip.c:3890 sip_write: Asked
> to transmit frame type 4, while native formats is 0x1000 (g722)
> (4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096)
>
> Is this a problem with 1.4.41 or my Polycom HD Voice phone with g722
> codec or both?
> (again - it works under 1.4.35 just prints a message many many times)
>
> Jerry
>
> --
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Re: [asterisk-users] asterisk 1.4.35 to 1.4.41

[Satish Patel]

Look like codec mismatch issue.

--
Sent from my iPhone

On May 3, 2011, at 9:55 PM, Jerry Geis  wrote:


Under 1.4.35 I get this message printed MANY times
[May  3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame  
type 4, while native formats is 0x1000 (g722)(4096) read/write =  
0x1000 (g722)(4096)/0x1000 (g722)(4096)
[May  3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame  
type 4, while native formats is 0x1000 (g722)(4096) read/write =  
0x1000 (g722)(4096)/0x1000 (g722)(4096)
[May  3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame  
type 4, while native formats is 0x1000 (g722)(4096) read/write =  
0x1000 (g722)(4096)/0x1000 (g722)(4096)
[May  3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame  
type 4, while native formats is 0x1000 (g722)(4096) read/write =  
0x1000 (g722)(4096)/0x1000 (g722)(4096)
[May  3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame  
type 4, while native formats is 0x1000 (g722)(4096) read/write =  
0x1000 (g722)(4096)/0x1000 (g722)(4096)


Under 1.4.41 I get an error and hang up doing the exact same thing.

All I am doing Is calling a cell phone over the PRI then dialing my  
SIP/524 extension.



This is from 1.4.35
 > Channel DAHDI/18-1 was answered.
  -- Executing [smvoice_callprogress@smvoice-dialout:1] GotoIf 
("DAHDI/18-1", "1?smvoice_callprogress|3:smvoice_callprogress|2") in  
new stack

  -- Goto (smvoice-dialout,smvoice_callprogress,3)
  -- Executing [smvoice_callprogress@smvoice-dialout:3] AGI("DAHDI/ 
18-1", "smvoice) in new stack

  -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
  -- Playing '/home/silentm/record/please_press/ 
one_to_call.' (escape_digits=0123456789*#) (sample_offset 0)
[May  3 21:47:38] DTMF[21746]: channel.c:2368 __ast_read: DTMF end  
'1' received on DAHDI/18-1, duration 0 ms
[May  3 21:47:38] DTMF[21746]: channel.c:2423 __ast_read: DTMF end  
accepted without begin '1' on DAHDI/18-1
[May  3 21:47:38] DTMF[21746]: channel.c:2434 __ast_read: DTMF end  
passthrough '1' on DAHDI/18-1
  -- Playing '/tmp/smvoice.21747_0' (escape_digits=0123456789#)  
(sample_offset 0)
[May  3 21:47:41] ERROR[21746]: utils.c:968 ast_carefulwrite: write 
() returned error: Broken pipe

  -- AGI Script smvoice completed, returning 0
  -- Executing [smvoice_dial_goto_voicemail@smvoice-dialout:1] Dial 
("DAHDI/18-1", "SIP/524|30|tT") in new stack

  -- Called 524
[May  3 21:47:41] WARNING[21746]: channel.c:3782  
ast_channel_make_compatible: No path to translate from SIP/ 
524-0001(4096) to DAHDI/18-1(4)
[May  3 21:47:41] WARNING[21746]: chan_sip.c:3890 sip_write: Asked  
to transmit frame type 4, while native formats is 0x1000 (g722) 
(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096)
[May  3 21:47:41] WARNING[21746]: chan_sip.c:3890 sip_write: Asked  
to transmit frame type 4, while native formats is 0x1000 (g722) 
(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096)


Is this a problem with 1.4.41 or my Polycom HD Voice phone with g722  
codec or both?

(again - it works under 1.4.35 just prints a message many many times)

Jerry

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[asterisk-users] Res: Fading voice problem

[Eduardo Leones]

10%




De: Matt Riddell 
Para: asterisk-users@lists.digium.com
Enviadas: Quarta-feira, 4 de Maio de 2011 0:32:28
Assunto: Re: [asterisk-users] Fading voice problem

On 3/05/11 10:16 PM, Eduardo Leones wrote:
> Guys,
>
> I'm having problems in the fading voice calls, receptive and active,
> that in SIP accounts. While few people using the system, calls are
> perfect, but it beats the normal use of connections (average 30
> concurrent), the voice begins to fade from people.
>
> Soon I figured some network problem, I did a tcpdump and analyzed by
> wireshark ...the strange thing is this ...
>
> all packets that arrive on the server asterisk are normal or jitter,
> latency ... But whenAsterisk sends packets to the network or the ISP ...
> maggoty packages are ... jitter of150ms on average ... latency of more
> than 1000 ms ...
>
> That is, by the way is not the network itself, but the network on the
> machine ...
>
> Dropped iptables to make sure no influence ... I changed the network
> card and cables... did nothing more ...
>
> Anyone have any ideas to help me and chase to find the problem?
>
> PS: The server is a CentOS 5.5 - 32 bit ... I've tested the 64bit tb but
> with the sameerror ...

What's your CPU usage like?

-- 
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[asterisk-users] Invalid use of undefined type when make dahdi

[CB]

I am attempting to install Dahdi on a virtual machine running Centos 5.5 and 
having various problems.

yum install kernel-devel gcc make gcc-c++ libxml2-devel
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
* base: mirror.optus.net
* extras: mirror.optus.net
* rpmforge: fr2.rpmfind.net
* updates: mirror.optus.net
Setting up Install Process
Package kernel-devel-2.6.18-238.9.1.el5.x86_64 already installed and
latest version
Package gcc-4.1.2-50.el5.x86_64 already installed and latest version
Package 1:make-3.81-3.el5.x86_64 already installed and latest version
Package gcc-c++-4.1.2-50.el5.x86_64 already installed and latest version
Package libxml2-devel-2.6.26-2.1.2.8.el5_5.1.x86_64 already installed
and latest version
Package libxml2-devel-2.6.26-2.1.2.8.el5_5.1.i386 already installed and
latest version
Nothing to do

[root@atlantis dahdi-linux-2.4.1.2]# make
make -C drivers/dahdi/firmware firmware-loaders
make[1]: Entering directory
`/usr/src/dahdi-linux-2.4.1.2/drivers/dahdi/firmware'
make[1]: Leaving directory
`/usr/src/dahdi-linux-2.4.1.2/drivers/dahdi/firmware'
You do not appear to have the sources for the 2.6.18-238.el5 kernel
installed.

So we're running a different kernel...

uname -r
2.6.18-238.el5

Downloaded kernel sources for 2.6.18-238.el5

[user@atlantis ~]$ mkdir -p ~/rpmbuild/{BUILD,RPMS,SOURCES,SPECS,SRPMS}
[user@atlantis ~]$ echo '%_topdir %(echo $HOME)/rpmbuild' >
~/.rpmmacros
[user@atlantis ~]$cd ~/rpmbuild/SPECS
[user@atlantis ~]$rpmbuild -bp --target=`uname -m` kernel-2.6.spec
2> prep-err.log | tee prep-out.log

[root@atlantis kernel-2.6.18]# cp
/home/user/rpmbuild/BUILD/kernel-2.6.18/linux-2.6.18.x86_64
/usr/src/kernels/2.6.18-238.el5-x86_64 -R

[root@atlantis dahdi-linux-complete-2.4.1.2+2.4.1]# make all

/usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:8652:
error: invalid use of undefined type ‘struct module’
/usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:8652:
error: ‘struct dahdi_chan’ has no member named ‘pulsecount’
etc etc

This article 
(http://asteriskfaqs.org/2011/01/30/asterisk-users/invalid-use-of-undefined-type-struct-module.html)
 indicates that those errors are the result of not having CONFIG_MODULES set in 
the kernel config. 

cd /home/user/rpmbuild/BUILD/kernel-2.6.18/linux-2.6.18.x86_64
make menuconfig
[*] Enable loadable module support

Legend: [*] built-in

Any advice appreciated.


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Re: [asterisk-users] How to debug MixMonitor misbehaviour

[salaheddine elharit]

hi

you can add this in extenssion.conf


exten => 223,1,Answer()

exten => 223,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))

exten => 223,3,Dial(SIP/223)

exten => 223,4,Hangup()

i can record without any issue in /var/spool/asterisk/monitor


2011/5/4 Bruce B 

> Thanks for the input.
>
> Yes, I did call out many times, but the recording doesn't happen even after
> the call is bridged and there is two way audio. I also took out the "b"
> option and so it should recording the ringing right (even before call is
> bridged) but it doesn't do that or any recording at all.
>
> Any other suggestions as to what I can do to see why this is not recording?
>
> Regards,
>
>
> On Tue, May 3, 2011 at 2:13 AM, virendra bhati  wrote:
>
>> Hi,
>>
>> As per your Dialplan MixMonitor will work after call bridge, In you case
>> still call is not bridge. That's why MixMonitor is waiting of call bridge...
>>
>> *
>> MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)
>> option b=>** A bridge flag allows recording to only take place when the
>> channel is bridged.*
>>
>> So just for test make sip call and start mixmonitor to test the recorded
>> file.
>> default path od recording id
>> *
>>  /var/spool/asterisk/monitor/
>>
>> *
>>  On Tue, May 3, 2011 at 10:40 AM, Bruce B  wrote:
>>
>>>  Hi everyone,
>>>
>>> For some reason MixMonitor doesn't record when it should; It actually
>>> shows the MixMonitor line just fine on the CLI. How can MixMonitor be
>>> debugged for things like privilege issues or filename issues?
>>>
>>> **I had this working at one point and then stopped working. Not sure what
>>> I changed.
>>>
>>> System Info:
>>> Asterisk 1.4.21.2
>>> Queuemetrics 1.6.3.0
>>>
>>>
>>> [queuedial]
>>>  ; this piece of dialplan is just a calling hook into the [qm-queuedial]
>>> context that actually does the
>>> ; outbound dialing - replace as needed - just fill in the same variables.
>>> exten => _XXX.,1,Set(QDIALER_QUEUE=q-${EXTEN:0:3})
>>> exten => _XXX.,n,Set(QDIALER_NUMBER=${EXTEN:3})
>>> exten => _XXX.,n,Set(QDIALER_AGENT=Agent/${CALLERID(num)})
>>> exten => _XXX.,n,Set(QDIALER_CHANNEL=ZAP/g0/${QDIALER_NUMBER})
>>> exten => _XXX.,n,Set(QueueName=${QDIALER_QUEUE})
>>> *exten => _XXX.,n,MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)*
>>> exten => _XXX.,n,Goto(qm-queuedial,s,1)
>>>
>>> CLI output:
>>>  -- Called 4904166356574@queuedial/n
>>> -- Executing [4904166356574@queuedial:1]
>>> Set("Local/4904166356574@queuedial-d851,2", "QDIALER_QUEUE=q-490") in
>>> new stack
>>> -- Executing [4904166356574@queuedial:2]
>>> Set("Local/4904166356574@queuedial-d851,2", "QDIALER_NUMBER=4166356574")
>>> in new stack
>>> -- Executing [4904166356574@queuedial:3]
>>> Set("Local/4904166356574@queuedial-d851,2",
>>> "QDIALER_AGENT=Agent/19053640558") in new stack
>>> -- Executing [4904166356574@queuedial:4]
>>> Set("Local/4904166356574@queuedial-d851,2",
>>> "QDIALER_CHANNEL=ZAP/g0/4166356574") in new stack
>>> -- Executing [4904166356574@queuedial:5]
>>> Set("Local/4904166356574@queuedial-d851,2", "QueueName=q-490") in new
>>> stack
>>> *-- Executing [4904166356574@queuedial:6]
>>> MixMonitor("Local/4904166356574@queuedial-d851,2",
>>> "Q-q-490-1304399098.18.WAV|b|") in new stack*
>>> -- Executing [4904166356574@queuedial:7]
>>> Goto("Local/4904166356574@queuedial-d851,2", "qm-queuedial|s|1") in new
>>> stack
>>> -- Goto (qm-queuedial,s,1)
>>>
>>> Trying to locate file:
>>>  root@pbx:~ $ updatedb
>>> root@pbx:~ $ locate Q-q-490-1304399098.18.WAV
>>> root@pbx:~ $ ls /var/spool/asterisk/monitor/Q-q*
>>> ls: /var/spool/asterisk/monitor/Q-q*: No such file or directory
>>>
>>> I also turned on the Debug but I couldn't see anything out of the norm.
>>> As you can see above the CLI output is just fine.
>>>
>>> Thanks,
>>> Bruce
>>>
>>> --
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>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>> --
>>
>>
>>
>> -
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-9172341457
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hel

Re: [asterisk-users] asterisk HA for queue calls

[DHAVAL INDRODIYA]

Hi Rajib,

I think It is not possible with asterisk , as primary server goes down it
will stop asterisk services so once asterisk service down i think all
connected calls to queue will hangup automatically, and you cannot retrive
those calls as they all are disconnected .

I think you need to consider more on load balancing per asterisk server in
that case the problem of Availability is solved to some level, If You using
SIP protocol then you can think of OPENSER and from that you can use
loadbalancer which routed calls in a way an depend on machine strength.

I hope this idea will useful to solve your requirement.

Regards
Dhaval

On Wed, May 4, 2011 at 1:13 PM, Deka, Rajib IN MAA SL <
rajib.d...@siemens.com> wrote:

>  Hello List,
>
>
>
> We are running two asterisk machines in virtual IP as primary and secondary
> server.
>
> Initially virtual IP will be active in primary server; during the failure
> of primary secondary will get the virtual IP.
>
>
>
> Is there any way to retrieve pending queue calls from primary to secondary,
> in case primary fails?
>
> Does asterisk provide any interface to do it or we have to write some
> application on asterisk to do the same.
>
>
>
> Regards,
>
> Rajib
>
>
>
> --
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> corporate proprietary information. If you have received it by mistake,
> please notify us immediately by reply e-mail and delete this e-mail and its
> attachments from your system.
> Thank You.
>
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> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] asterisk HA for queue calls

[Deka, Rajib IN MAA SL]

Hello List,

We are running two asterisk machines in virtual IP as primary and secondary 
server.
Initially virtual IP will be active in primary server; during the failure of 
primary secondary will get the virtual IP.

Is there any way to retrieve pending queue calls from primary to secondary, in 
case primary fails?
Does asterisk provide any interface to do it or we have to write some 
application on asterisk to do the same.

Regards,
Rajib



Important notice: This e-mail and any attachment there to contains corporate 
proprietary information. If you have received it by mistake, please notify us 
immediately by reply e-mail and delete this e-mail and its attachments from 
your system.
Thank You.
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Re: [asterisk-users] Multiple cards using same IRQ - getting IRQ errors and hissing

[Johan Wilfer]

On 2011-05-03 16:32, Dean Hoover wrote:

I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an
HP ML110 G6 using Ubuntu Linux 10.04 LTS.

I have two Digium TE121 single T1 port cards and a Digium AEX800
8-port FXS card.  All PCI Express cards.

Co-workers are hearing hissing sounds on some calls, and I am getting
IRQ errors when running "dahdi show status".

I see that sharing IRQs for Digium cards isn't recommended, so I'm
trying to set it so each card gets its own.  From the few web sites
I've read so far, including Digium's FAQ site, I've added ACPI and
verified that the BIOS does not give me the ability to manually set
the IRQ.  I've even taken one of the TE121's out of the server (it
isn't being used anyways).  Everything I've done so far has not fixed
it.  All the cards (as well as USB1) all use IRQ 16.

The other option given was to use setpci, but I am unfamiliar with
that command.  I did what I could to try and find the setting (based
on what the man page on Ubuntu's web site) where I could see the value
16, but not getting anywhere.

I know that this is more of an Asterisk forum than Digium.  If I need
to put in a case at Digium I will, but wanted to see if there were any
suggestions here before I pursued that.

Any help would be appreciated.

Dean Hoover



A month ago I had similar problems with a HP DL360g6 and a HP DL380g7 
running Debian 5 "Lenny".
In the HP DL360g6 I had one TE121. I noticed IRQ misses and the problem 
was easily reproduced
by running dahdi_maint to enable loopback and patlooptest while 
compiling asterisk to create some i/o.


When I installed Debian 6 "Squeeze" instead the problem went away. 
Tested with both servers above.
On this page I found some information about APIC (Advanced Programmable 
Interupt Controller)
http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html 
(quite old but informative)


I haven't got the time to verify the root cause of the problem yet (I've 
planned to do this at the end of this month)
but my theory is that it has something to do with the kernels APIC 
handling that was fixed between Debian 5 and 6.


Maybe you experience something similar?

/Johan

--
Johan Wilfer email: jo...@jttech.se
JT Tech | Utvecklare webb: http://jttech.se
direkt: +46 31 380 91 01  support: +46 31 380 91 00


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