[asterisk-users] Detect Low Quality Calls - Realtime

[XBrian]

Hi there,
I support a large number of enterprise users who contractually must connect to 
our support center via a 4G VOIP connection.

I simply want to be able to auto detect all poor quality calls in realtme (as 
they are being made), play a message and drop the call - without user 
intervention. All decent call quality calls will be allowed through - to be 
handled by support staff.

Its a challenging and tricky one as I cannot install any software on the 
callers 
endpoint. I can only detect calls as they hit our server, do the magic and 
based 
on latency, bandwidth and MOS (Meaning Opinion Score)  - decide whether the 
call 
should be let through. I will accept all MOS values of 4.0

Any bright ideas?


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Re: [asterisk-users] [Spam] Re: Calender and EWS with shared calenders

[Magnus Löfqvist]

4 jan 2013 kl. 23:59 skrev "Danny Nicholas" 
mailto:da...@debsinc.com>>:


From: 
asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magnus Löfqvist
Sent: Friday, January 04, 2013 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Calender and EWS with shared calenders

Hi all,

We want to use res_calender_ews to close users extensions if there are busy in 
there exchange calenders.

It is possible to use shared calenders ? Ie, I have a resource user that have 
access to the users calenders, so I dont need to maintain every users 
password/username in asterisk.

We are today running 1.8 but are going to upgrade to 11...

Best regards / Magnus

If it works for you in 1.8 it should work in 11.X – the only change is the 
$Revision number.


Dont know if it workes in 1.8. Havent set it up yet.
My question is if it works in 1.8 (or if it will work in 11)

/ Magnus
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Re: [asterisk-users] Calender and EWS with shared calenders

[Danny Nicholas]

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magnus
Löfqvist
Sent: Friday, January 04, 2013 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Calender and EWS with shared calenders

 

Hi all, 

 

We want to use res_calender_ews to close users extensions if there are busy
in there exchange calenders.

 

It is possible to use shared calenders ? Ie, I have a resource user that
have access to the users calenders, so I dont need to maintain every users
password/username in asterisk.

 

We are today running 1.8 but are going to upgrade to 11...

 

Best regards / Magnus

 

If it works for you in 1.8 it should work in 11.X – the only change is the
$Revision number.

 

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[asterisk-users] Calender and EWS with shared calenders

[Magnus Löfqvist]

Hi all,

We want to use res_calender_ews to close users extensions if there are busy in 
there exchange calenders.

It is possible to use shared calenders ? Ie, I have a resource user that have 
access to the users calenders, so I dont need to maintain every users 
password/username in asterisk.

We are today running 1.8 but are going to upgrade to 11...

Best regards / Magnus

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Re: [asterisk-users] Unable to build DAHDI

[Russ Meyerriecks]

Looks like this issue:

https://issues.asterisk.org/jira/browse/DAHTOOL-60

-- 
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Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6025
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] Unable to build DAHDI

[Bruce Ferrell]

Hi all,

I'm attempting to install dahdi-linux-complete-2.6.1+2.6.1 on an OpenSUSE 
system running kernel 3.7.1-17

The build is failing like this:

ct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tone_detection.c oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsi_cnct.c 
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsst.c oct612x/apilib/bt/octapi_bt0.c oct612x/apilib/largmath/octapi_largmath.c oct612x/apilib/llman/octapi_llman.c

oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_conf_bridge.c: In function 
‘Oct6100ApiBridgeEventRemove’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_conf_bridge.c:3870:47: 
error: ‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_conf_bridge.c:3870:47: 
note: each undeclared identifier is reported only once for each function it 
appears in
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c: In function 
‘Oct6100EventGetToneDef’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c:89:32: error: 
‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c:89:32: note: each 
undeclared identifier is reported only once for each function it appears in
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c: In function 
‘Oct6100BufferPlayoutTransferEvents’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c:1116:126: error: 
‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c: In function 
‘Oct6100BufferPlayoutCheckForSpecificEvent’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c:1287:29: error: 
‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_miscellaneous.c: In 
function ‘Oct6100ApiStrStr’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_miscellaneous.c:391:38: 
error: ‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_miscellaneous.c:391:38: 
note: each undeclared identifier is reported only once for each function it 
appears in
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_playout_buf.c: In function 
‘Oct6100BufferPlayoutStopDef’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_playout_buf.c:530:43: 
error: ‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_playout_buf.c:530:43: note: 
each undeclared identifier is reported only once for each function it appears in
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_playout_buf.c: In function 
‘Oct6100ApiInvalidateChanPlayoutStructs’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_playout_buf.c:2844:51: 
error: ‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsst.c: In function 
‘Oct6100ApiReserveTsst’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsst.c:401:29: error: 
‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsst.c:401:29: note: each 
undeclared identifier is reported only once for each function it appears in
oct612x/apilib/bt/octapi_bt0.c: In function ‘OctApiBt0Init’:
oct612x/apilib/bt/octapi_bt0.c:63:12: error: ‘NULL’ undeclared (first use in 
this function)
oct612x/apilib/bt/octapi_bt0.c:63:12: note: each undeclared identifier is 
reported only once for each function it appears in
oct612x/apilib/llman/octapi_llman.c: In function ‘OctapiLlmAllocInit’:
oct612x/apilib/llman/octapi_llman.c:88:12: error: ‘NULL’ undeclared (first use 
in this function)
oct612x/apilib/llman/octapi_llman.c:88:12: note: each undeclared identifier is 
reported only once for each function it appears in
make[3]: *** [oct6100_adpcm_chan.o] Error 1
make[3]: Leaving directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.6.1+2.6.1/tools/xpp'
make[2]: *** [utils-subdirs] Error 2
make[2]: Leaving directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.6.1+2.6.1/tools'
make[1]: *** [all] Error 2
make[1]: Leaving directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.6.1+2.6.1/tools'
make: *** [all] Error 2


Any suggestions are welcome



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Re: [asterisk-users] faxdetect on/off on the fly?

[Steve Edwards]

On Fri, 4 Jan 2013, Danny Nicholas wrote:


Simple Perl AGI to set dialplan variable:

print STDOUT "SET VARIABLE USEFAX \"ON\" \n";

print STDOUT "SET VARIABLE USEFAX \"OFF\" \n";


You need to read the response from each request to comply with the AGI 
protocol.


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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] faxdetect on/off on the fly?

[Eric Wieling]

I believe Asterisk 11 is the first version which allows you to enable and 
disable faxdetect on the fly.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Friday, January 04, 2013 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] faxdetect on/off on the fly?

On Thu, Jan 3, 2013 at 9:39 PM, David Cunningham  
wrote:


We have all calls going to an AGI, which decides where the number will 
get routed to, and if fax detection should be enabled for this call. The choice 
should only apply to the current call.



What criteria would determine if fax detection should be enabled?  From reading 
this message, what it sounds like is you want the call to go to the AGI, and if 
a CNG tone is detected, you want it to go to a specific fax extension.  That's 
what faxdetect does.  You enable it on all your lines, and if a CNG tone is 
detected, it sends it to the "fax" exten in the current context.  This would 
remove your routing AGI form the picture, so I don't think you want faxdetect 
enabled on your lines.  


Maybe I'm misunderstanding, but to me, it seems like you're trying to detect a 
CNG tone and base your routing decision on that inside your AGI.  Faxdetect 
will detect the CNG tone after the call is answered and automatically route for 
you.  It's not the kind of thing you want to set on a call by call basis.  If 
you're looking to detect a CNG tone inside your AGI, I'm not sure what 
mechanism is available for that.  



--Warren Selby, dCAP
http://www.SelbyTech.com


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Re: [asterisk-users] faxdetect on/off on the fly?

[Warren Selby]

On Thu, Jan 3, 2013 at 9:39 PM, David Cunningham
wrote:

> We have all calls going to an AGI, which decides where the number will get
> routed to, and if fax detection should be enabled for this call. The choice
> should only apply to the current call.
>

What criteria would determine if fax detection should be enabled?  From
reading this message, what it sounds like is you want the call to go to the
AGI, and if a CNG tone is detected, you want it to go to a specific fax
extension.  That's what faxdetect does.  You enable it on all your lines,
and if a CNG tone is detected, it sends it to the "fax" exten in the
current context.  This would remove your routing AGI form the picture, so I
don't think you want faxdetect enabled on your lines.

Maybe I'm misunderstanding, but to me, it seems like you're trying to
detect a CNG tone and base your routing decision on that inside your AGI.
Faxdetect will detect the CNG tone after the call is answered and
automatically route for you.  It's not the kind of thing you want to set on
a call by call basis.  If you're looking to detect a CNG tone inside your
AGI, I'm not sure what mechanism is available for that.


--Warren Selby, dCAP
http://www.SelbyTech.com 
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Re: [asterisk-users] faxdetect on/off on the fly?

[Danny Nicholas]

Don't think you actually can, per se.  What you can do is set a variable and
redirect to the line that has this defined or undefined.

Let's say that you have 4 lines;  SIP/1001 and DAHDI/1 have faxdetect=yes
defined in sip.conf and chan_dahdi.conf. SIP/1002 and DAHDI/2 have
faxdetect=no in the respective places.  Simple Perl AGI to set dialplan
variable:

cat selline.pl

#!/usr/local/bin/perl

#

#

# Send USEFAX to Dialplan

#

#

use strict;

use warnings;

require Asterisk::AGI;

# turn off I/O buffering

$| = 1;

 

# the AGI object

my $agi = new Asterisk::AGI;

my %input = $agi->ReadParse();

 

print STDOUT "SET VARIABLE USEFAX \"ON\" \n";

print STDOUT "SET VARIABLE USEFAX \"OFF\" \n";

exit;

In the Dialplan

Exten => s,1,Answer()

Exten => s,n,AGI(selline.pl)

Exten => s,n,Gotoif($

exten => s,1,Gotoif($[${USEFAX} = ON]?on:off)

exten => s,n(on),Dial(SIP/1001)

exten => s,n,hangup

exten => s,n(off),Dial(SIP/1002)

exten => s,n,hangup

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Friday, January 04, 2013 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] faxdetect on/off on the fly?

 

Hi Danny,

Can you please elaborate on how in the dialplan we can set faxdetect on and
off?

We currently have it set on in sip.conf.

Thanks.



On 3 January 2013 17:21, Danny Nicholas  wrote:

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Thursday, January 03, 2013 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] faxdetect on/off on the fly?

 

Hello,

We want the ability to choose from an AGI script whether or not to enable
faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone
suggest a workaround?

Thanks for any advice.


You should be able to call the AGI and set a dialplan variable and use
Gotoif to do/not do faxdetect.  Reading the .sample files for 11.0 it seems
that normally these are "configured until restart/reload" but with a little
testing, the default should be overrideable.


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-- 
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019

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Re: [asterisk-users] Verizon SIP "trunking" Field Trial

[Michael L. Young]

- Original Message -
> From: "Matthew J. Roth" 

> At least Verizon maintains a consistent customer experience.  ; )
> 
> Overall, we've found the service to be reliable and stable, but when
> there are problems or changes needed you're dealing with Verizon and
> the
> w...h...e...e...l...s..t...u...r...n..s...l...o...w...l...y.

Haha... that is funny... it is sooo true.

Well, you are right.  Once it is working, it is usually pretty stable.  Just a 
pain in the butt when things are not working.  Hopefully we can get through the 
Field Trial and that is all I have to worry about for a while.

Thanks Matthew for all the encouragement as I go down this temporary (I hope) 
unpleasant path.

Michael

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Re: [asterisk-users] Verizon SIP "trunking" Field Trial

[Michael L. Young]

- Original Message - 
> From: "Carlos Alvarez" 

> Sounds like the same huge effort it takes to work with
> Qwest/Centurylink, and in the long run we found it simply isn't
> worth it. The few benefits of working with an RBOC are countered by
> the many drawbacks of working with an RBOC.

> Also we recently acquired a half million minutes/mo from a company
> who was tired of dealing with Qwest SIP. They said the same thing I
> said above.

> I suppose the point of what I'm saying is you should really think
> about what you think you will gain from a relationship with them,
> and whether all this is worth it ("all this" means now and how their
> attitude will affect you forever).

Trust me, this was not my choice... They are not fun to deal with when it came 
to our PRI lines.  After dealing with dropped calls and errors on the T1s, they 
wouldn't admit that they had a problem until finally they looked at the 
hardware at the CO while we were down hard (which cost us about 4 - 6 hours 
downtime) and said, "Oh, we do have a problem".  To make a long story short, it 
was fixed and has been good since but I was really trying to move us away from 
Verizon.  Unfortanately, it boils down to cost and Verizon being as big as they 
are were able to make a deal (getting us out of contracts that had been signed, 
credits, etc.) that the ultimate decision maker here at the company went for.  
That decision maker also has the mindset that we have to stick with the phone 
company for some reason.  I was strongly against it and wanted to go with a 
different company.  So, I have to deal with it now.

Thanks for your input.  It pretty much echoes my sentiments.

Michael

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Re: [asterisk-users] Verizon SIP "trunking" Field Trial

[Matthew J. Roth]

> Carlos Alvarez wrote:
>
> Sounds like the same huge effort it takes to work with Qwest/
> Centurylink, and in the long run we found it simply isn't worth it.
> The few benefits of working with an RBOC are countered by the many
> drawbacks of working with an RBOC.
> 
> Also we recently acquired a half million minutes/mo from a company who
> was tired of dealing with Qwest SIP.  They said the same thing I said
> above.
> 
> I suppose the point of what I'm saying is you should really think
> about what you think you will gain from a relationship with them, and
> whether all this is worth it ("all this" means now and how their
> attitude will affect you forever).
> 
> 
>> Eric Wieling wrote:
>> 
>> Trust me, Verizon doesn't really provide support.What they will
>> do is tell you something different (often conflicting stuff) when you
>> send in a ticket.One time they tell us the From must be in e.164
>> format, other times they say it does not.We asked for an updated
>> Interop guide weeks ago and they have not provided us anything.  We
>> have been with VZ SIP for years so I wanted an updated interop guide
>> so we can point them to it when they tell us something which
>> conflicts with their docs.  Don't get me started on trying to upgrade
>> our service with them..


Michael,

At least Verizon maintains a consistent customer experience.  ; )

Overall, we've found the service to be reliable and stable, but when
there are problems or changes needed you're dealing with Verizon and the
w...h...e...e...l...s..t...u...r...n..s...l...o...w...l...y.

It's a trade-off of dealing with such a huge company and I don't have
experience with any other SIP trunk providers to compare it to, so give
proper consideration to the opinions of the other members of this list.

Just know that as you weigh your options, Verizon will work in the end
but this whole field trial process is your preview of all future
dealings with them.  It can make you question your sanity but so far
everything you've said has been spot-on.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] faxdetect on/off on the fly?

[David Cunningham]

Hi Danny,

Can you please elaborate on how in the dialplan we can set faxdetect on and
off?

We currently have it set on in sip.conf.

Thanks.


On 3 January 2013 17:21, Danny Nicholas  wrote:

> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Cunningham
> *Sent:* Thursday, January 03, 2013 3:13 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] faxdetect on/off on the fly?
>
> ** **
>
> Hello,
>
> We want the ability to choose from an AGI script whether or not to enable
> faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone
> suggest a workaround?
>
> Thanks for any advice.
>
> You should be able to call the AGI and set a dialplan variable and use
> Gotoif to do/not do faxdetect.  Reading the .sample files for 11.0 it seems
> that normally these are “configured until restart/reload” but with a little
> testing, the default should be overrideable.
>
>
> --
> _
> -- 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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USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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Re: [asterisk-users] Verizon SIP "trunking" Field Trial

[Carlos Alvarez]

On Fri, Jan 4, 2013 at 11:18 AM, Eric Wieling  wrote:

> Trust me, Verizon doesn't really provide support.What they will do is
> tell you something different (often conflicting stuff) when you send in a
> ticket.One time they tell us the From must be in e.164 format, other
> times they say it does not.We asked for an updated Interop guide weeks
> ago and they have not provided us anything.  We have been with VZ SIP for
> years so I wanted an updated interop guide so we can point them to it when
> they tell us something which conflicts with their docs.  Don't get me
> started on trying to upgrade our service with them..
>

Sounds like the same huge effort it takes to work with Qwest/Centurylink,
and in the long run we found it simply isn't worth it.  The few benefits of
working with an RBOC are countered by the many drawbacks of working with an
RBOC.

Also we recently acquired a half million minutes/mo from a company who was
tired of dealing with Qwest SIP.  They said the same thing I said above.

I suppose the point of what I'm saying is you should really think about
what you think you will gain from a relationship with them, and whether all
this is worth it ("all this" means now and how their attitude will affect
you forever).


-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Verizon SIP "trunking" Field Trial

[Eric Wieling]

Trust me, Verizon doesn't really provide support.What they will do is tell 
you something different (often conflicting stuff) when you send in a ticket.
One time they tell us the From must be in e.164 format, other times they say it 
does not.We asked for an updated Interop guide weeks ago and they have not 
provided us anything.  We have been with VZ SIP for years so I wanted an 
updated interop guide so we can point them to it when they tell us something 
which conflicts with their docs.  Don't get me started on trying to upgrade our 
service with them..

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael L. Young
Sent: Friday, January 04, 2013 1:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Verizon SIP "trunking" Field Trial

- Original Message -
> From: "Matthew J. Roth" 

> Your email documents the same experience we had years ago.  It was 
> strange reading it and I was shocked that nothing has changed in that 
> much time.  Asterisk will work with Verizon's IP trunking product, but 
> they're trying to make you jump through some old hoops first.

Those were my thoughts.  They are making this a lot more complicated than it 
really needs to be.  I think the main thing they are worried about is having to 
support something they don't know anything about.  Well, they won't have to 
support it; we will.  Just as long as they are SIP compliant, there should be 
no issues.

> We were using Verizon IP trunks over an MPLS network in 2008.  At the 
> time, they did not require IPSEC for signaling.  However, they did 
> want us to install an SBC and actually provided us with an AudioCodes 
> nCite 1000 at their cost.  It just acted as a proxy, so it didn't 
> affect interoperability with Verizon's IP trunks and I wouldn't buy 
> one only to satisfy them.

One of the engineers stated that they have received the direction to only use 
"standard" equipment.  So, they are afraid that our setup will not pass ICB 
since it doesn't fit into their "standard" way of doing things.

> We were quite happy with the service, so I'd encourage you to go ahead 
> with the field trial without putting an SBC in place.  Remember that 
> you will be paying them, so they should be working to fit your design 
> and if they reject you for some arcane reason then you are better off 
> with another provider anyway.
> 
> Don't hesitate to let them know that you know you're jumping through 
> the same hoops that have been in place since 2008 and you'd appreciate 
> it if they would streamline the process to save time and money.  Tell 
> them that Asterisk should already be on their certified list of 
> approved devices because they've been running field trials and 
> production setups on it for years.

It is good to hear that you were happy with the service.  I have my 
reservations with all the hoops they are making us jump through and that gives 
me a bit of confidence that it will be worth it.

I did tell them that Asterisk is being used all over the place as well as in 
big call centers.  I know that I have seen others in the Asterisk community on 
Verizon.  Verizon seems to be hung up on this certification stuff and it is 
hard to explain to them that this is not a piece of hardware you buy and 
plugin.  We can build our own servers and put Asterisk on it, and they seem to 
cringe when they hear that.

Thanks for your input Matthew.  It is appreciated.

Michael

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Re: [asterisk-users] Verizon SIP "trunking" Field Trial

[Michael L. Young]

- Original Message -
> From: "Matthew J. Roth" 

> Your email documents the same experience we had years ago.  It was
> strange reading it and I was shocked that nothing has changed in that
> much time.  Asterisk will work with Verizon's IP trunking product,
> but
> they're trying to make you jump through some old hoops first.

Those were my thoughts.  They are making this a lot more complicated than it 
really needs to be.  I think the main thing they are worried about is having to 
support something they don't know anything about.  Well, they won't have to 
support it; we will.  Just as long as they are SIP compliant, there should be 
no issues.

> We were using Verizon IP trunks over an MPLS network in 2008.  At the
> time, they did not require IPSEC for signaling.  However, they did
> want us to install an SBC and actually provided us with an AudioCodes
> nCite 1000 at their cost.  It just acted as a proxy, so it didn't
> affect interoperability with Verizon's IP trunks and I wouldn't
> buy one only to satisfy them.

One of the engineers stated that they have received the direction to only use 
"standard" equipment.  So, they are afraid that our setup will not pass ICB 
since it doesn't fit into their "standard" way of doing things.

> We were quite happy with the service, so I'd encourage you to go
> ahead
> with the field trial without putting an SBC in place.  Remember that
> you will be paying them, so they should be working to fit your design
> and if they reject you for some arcane reason then you are better off
> with another provider anyway.
> 
> Don't hesitate to let them know that you know you're jumping through
> the same hoops that have been in place since 2008 and you'd
> appreciate
> it if they would streamline the process to save time and money.  Tell
> them that Asterisk should already be on their certified list of
> approved devices because they've been running field trials and
> production setups on it for years.

It is good to hear that you were happy with the service.  I have my 
reservations with all the hoops they are making us jump through and that gives 
me a bit of confidence that it will be worth it.

I did tell them that Asterisk is being used all over the place as well as in 
big call centers.  I know that I have seen others in the Asterisk community on 
Verizon.  Verizon seems to be hung up on this certification stuff and it is 
hard to explain to them that this is not a piece of hardware you buy and 
plugin.  We can build our own servers and put Asterisk on it, and they seem to 
cringe when they hear that.

Thanks for your input Matthew.  It is appreciated.

Michael

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Re: [asterisk-users] Verizon SIP "trunking" Field Trial

[Michael L. Young]

> From: "Carlos Alvarez" 

> It may be too late for this, but in working with another RBOC who
> didn't want to deal with Asterisk, I just asked what they do
> support, and modified the headers sent by Asterisk to claim that it
> was one of the devices on that list. Done.

Like everyone else, I was laughing as well when I read this.

One engineer stated that they like to have an SBC to manipulate the headers to 
normalize things.  I stated that Asterisk was capable of manipulating headers 
if need be.  You just proved that it works :)

Michael

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Re: [asterisk-users] Polycom IP6000 upgrading and looping

[Kevin Larsen]

Justin,

I haven't seen it on that model, but I did have a case awhile back where 
it happened to me with a different conference phone. Pretty much the same 
symptoms you had. Even more fun it was remote so I couldn't get my hands 
on it.

I tracked mine down to being an incorrect firmware for that phone. I 
downloaded the recommended firmware for my phone and placed it in a 
subdirectory of my ftp server. Then I deleted the link to the old firmware 
for that model only and relinked it to the new firmware.

There are two pieces of firmware you might need to look at. There is the 
bootrom.ld file for your particular phone and then the sip_xxx.ld file for 
your particular phone. Either one of them could be the problem. In my 
case, the phone was updating the -app.log and -boot.log files, so I did have some clues as  to where it was 
running off the rails. 

I was able to modify the .cfg file to point only the 
conference phone to the different firmware so that I could still keep the 
other phones on the known working firmware. 

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208--
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[asterisk-users] Asterisk + Huawei K3765

[Jonson Player]

Hello,

I want to use an Huawei stick model K3765 which support voice with
asterisk. I'm begginer with this kind of interaction from asterisk
with external devices.
Can someone guide me what should i configure to use this device?

Thank you for support,

Regards,
Jonson.

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Re: [asterisk-users] MaxCallBR Peer Setting

[XBrian]

Its so obvious now that you've made it clear

Thanks


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[asterisk-users] T38MaxBitRate issue on fax passthrough

[Kevin Larsen]

Having an issue with receiving faxes, but when I pass through the fax.

Currently, I receive the fax with Digium's Fax for Asterisk, store it and 
the initiate an outbound call to our fax server. (XMedius Fax). This 
works, but we would prefer to have Asterisk simply route the call directly 
to the fax server and take the store and forward out of the equation. 

When I do that, however, the fax is never properly negotiated. One thing I 
have noticed is that XMedius Fax tells Asterisk it has 
'a=T38maxBitRate:14400' and Asterisk immediately turns around and tells 
our upstream provider 'a=T38MaxBitRate:2400' on the invites (full invite 
text below). Is the fact that XMedius is not capitalizing the 'm' in 
'T38maxBitRate' the cause of Asterisk telling the upstream provider that 
the 'T38MaxBitRate' is 2400?

This should be the relevant sip debug. I have replaced the IP addresses 
with XXX.XXX.XXX.XXX (or WWW or YYY or ZZZ) as appropriate.
<--- SIP read from UDP:WWW.WWW.WWW.WWW:5060 --->
INVITE sip:4803836...@zzz.zzz.zzz.zzz:5060 SIP/2.0
Via: SIP/2.0/UDP WWW.WWW.WWW.WWW:5060;branch=z9hG4bK-40A939C7AE8C
From: sip:XMFAX2.mydomain.world;tag=095775A0931E
To: sip:4803836...@zzz.zzz.zzz.zzz;tag=as09ca5622
Call-ID: 176a274d5342aac505d0125979d19...@zzz.zzz.zzz.zzz:5060
CSeq: 103 INVITE
Max-Forwards: 70
Contact: sip:WWW.WWW.WWW.WWW:5060
User-Agent: XMediusFAX/7.0.0.298
Content-Type: application/sdp
Content-Length: 315

v=0
o=XMedius-Fax-Gateway 76811410 411 IN IP4 WWW.WWW.WWW.WWW
s=Asterisk PBX 10.5.0-digiumphones
c=IN IP4 WWW.WWW.WWW.WWW
t=0 0
m=image 54296 udptl t38
a=T38FaxVersion:0
a=T38maxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:8192
a=T38FaxMaxDatagram:1008
a=T38FaxUdpEC:t38UDPRedundancy
<->
--- (11 headers 12 lines) ---
Sending to WWW.WWW.WWW.WWW:5060 (no NAT)
  == Using UDPTL CoS mark 5
Got T.38 offer in SDP in dialog 
176a274d5342aac505d0125979d19...@zzz.zzz.zzz.zzz:5060
Capabilities: us - (ulaw), peer - 
audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 
(nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.

<--- Transmitting (no NAT) to WWW.WWW.WWW.WWW:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
WWW.WWW.WWW.WWW:5060;branch=z9hG4bK-40A939C7AE8C;received=WWW.WWW.WWW.WWW
From: sip:XMFAX2.mydomain.world;tag=095775A0931E
To: sip:4803836...@zzz.zzz.zzz.zzz;tag=as09ca5622
Call-ID: 176a274d5342aac505d0125979d19...@zzz.zzz.zzz.zzz:5060
CSeq: 103 INVITE
Server: Asterisk PBX 10.5.0-digiumphones
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Contact: 
Content-Length: 0


<>
  == Using UDPTL CoS mark 5
set_destination: Parsing  for 
address/port to send to
set_destination: set destination to XXX.XXX.XXX.XXX:5060
Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:5060:
INVITE sip:4803836...@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK7ef2185a;rport
Max-Forwards: 70
From: ;tag=as40d4ca92
To: ;tag=gK020b0efc
Contact: 
Call-ID: 335684047_12245...@xxx.xxx.xxx.xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.5.0-digiumphones
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 505811356 505811358 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=Asterisk PBX 10.5.0-digiumphones
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=image 4464 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:2400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:507
a=T38FaxUdpEC:t38UDPFEC

---

<--- SIP read from UDP:XXX.XXX.XXX.XXX:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK7ef2185a;rport=5060
From: ;tag=as40d4ca92
To: ;tag=gK020b0efc
Call-ID: 335684047_12245...@xxx.xxx.xxx.xxx
CSeq: 102 INVITE
Content-Length: 0

<->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:XXX.XXX.XXX.XXX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK7ef2185a;rport=5060
From: ;tag=as40d4ca92
To: ;tag=gK020b0efc
Call-ID: 335684047_12245...@xxx.xxx.xxx.xxx
CSeq: 102 INVITE
Accept: application/sdp, application/isup, application/dtmf, 
application/dtmf-relay, multipart/mixed
Contact: 
Allow: 
INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Supported: timer
Session-Expires: 1800;refresher=uas
Content-Length: 303
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 28889 23043 IN IP4 XXX.XXX.XXX.XXX
s=SIP Media Capabilities
c=IN IP4 208.49.73.36
t=0 0
m=image 25030 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:2400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:262
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv
<-

[asterisk-users] Polycom IP6000 upgrading and looping

[Justin Sherrill]

I have a Polycom IP6000 conference phone, along with a lot of Polycom IP550 
units.  I've been updating all the 650s to Polycom's from 3.2.3  to the 4.0.3 
software release, by hodling 468*and having them pull the update.  

It's been fine with the 650s, but the IP6000 (held 68* for that one) keeps 
going in a loop - downloads updater, saves it, formats the filesystem, 
downloads the new bootROM, and then repeats.  There's no error on screen and no 
successful upload of logs to show an error.

Has anyone updated these models before and seen this?

Justin Sherrill - American Rock Salt
P: 585-991-6825 F: 585-991-6925 C: 585-298-6826


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Re: [asterisk-users] MaxCallBR Peer Setting

[Bharat Lalcheta]

Its maximum call Bit rate available for that peer. Default is 384 kbps.
Your call for that peer allowed max bit rate or bandwidth of 384 kpbs only

Regards,

Bharat Lalcheta
On Fri, Jan 4, 2013 at 7:09 PM, XBrian  wrote:

> Hi
>
> sip show peer 21342
>
> gives me peer 21342's parameters. I am interested in the MaxCallBR line
> i.e.
>
>   MaxCallBR: 384 kbps
>
>
> What exactly does this mean?
>
>
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[asterisk-users] MaxCallBR Peer Setting

[XBrian]

Hi 

sip show peer 21342

gives me peer 21342's parameters. I am interested in the MaxCallBR line i.e.

  MaxCallBR: 384 kbps


What exactly does this mean?


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[asterisk-users] WebM / VP8 support

[Marek Cervenka]

hello,

any news about WebM/VP8 support in asterisk?
some bounty where can i contribute?



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===

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Re: [asterisk-users] Moving User Agent To Remote Location

[isrlgb]

Did you set externip and localnet in your sip conf ?


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Re: [asterisk-users] Moving User Agent To Remote Location

[Ishfaq Malik]

On Thu, 2013-01-03 at 12:47 -0500, Nick Khamis wrote:
> Hello Everyone,
> 
> Before getting into SIP and RTP traces, I wanted to clarify some of
> the sip.conf settings that may to some seem redundant or have a
> misconception with. I do apologize if this has been discussed time and
> time again as I would imagine. If anything, this email would make
> google search results that much stronger :).
> 
> With the UA local to my network I had tested two way audio, and now
> with the phone outside of network, we have no way audio. Before
> discussing NAT (which is enabled on the peer), and port forwarding
> (which is setup on the remote location), I would like to make sure I
> fully understand all the sip.conf settings. We are using Asterisk
> realtime via sip_buddies, and the fields in question are:
> 
> (Enclosed in brackets are an example value for the setting)
> 
> * host (dynamic): No problem here. Just wanted to mention that it's
> set as such
> * nat (yes): No problem here either
> * defaultuser (1...@example.com): Does the "@example.com" have to
> point to the UA (i.e., (1003@ua-public-ip), or is it just a name type
> field?
> * fullcontact: What to put here for a UA that is running at a remote
> location with dynamic external IP?
> * ipaddr (ua-public-ip): I did try setting it to the public ip of the
> UA, but is that really practical?
> What if I don't know where the initial registration request is coming
> from? I am guessing "host=dynamic" takes care of that.
> * defaultip??
> * dynamic: Should this be set to yes, or is "host=dynamic" sufficient?
> 
> The phone registers fine, and terminates a call through our providers.
> Just no audio both ways, which would suggest something more that an
> RTP issue which should at least have one way outgoing audio.
> 
> Things that have been attempted:
> * Port forwarding to the phone
> * Changing defaultuser to 1003@ua-public-ip. This made our OpenSIPS
> sip proxy through a fit.
> 
> Things I will attempt today:
> Calling the UA extension from an extension here
> SIP trace
> 
> Your help is greatly appreciated!!!
> 
> Nick.
> 

Hi

Is your directmedia/canreinvite (depending on asterisk version) set to
no?

Regards

Ish

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Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
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Re: [asterisk-users] User busy issue in A400P 4 FXO card

[A J Stiles]

On Friday 04 January 2013, Selva M wrote:
> Hi,
> 
>   I tried the option and got following message.
> 
> PBX1*CLI>
> -- Starting simple switch on 'DAHDI/1-1'
>   == Starting DAHDI/1-1 at from-pstn,s,1 failed so falling back to exten
> 's' == Starting DAHDI/1-1 at from-pstn,s,1 still failed so falling back to
> context 'default'
> -- Executing [s@default:1] Playback("DAHDI/1-1", "vm-goodbye") in new
> stack

Your Asterisk either is not seeing the [from-pstn] context, or is including 
another context called [from-pstn].  Check that this context occurs only once 
in extensions.conf, and is spelt the same way in extensions.conf and in 
chan_dahdi.conf.



-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Moving User Agent To Remote Location

[Ishfaq Malik]

On Thu, 2013-01-03 at 12:47 -0500, Nick Khamis wrote:
> Hello Everyone,
> 
> Before getting into SIP and RTP traces, I wanted to clarify some of
> the sip.conf settings that may to some seem redundant or have a
> misconception with. I do apologize if this has been discussed time and
> time again as I would imagine. If anything, this email would make
> google search results that much stronger :).
> 
> With the UA local to my network I had tested two way audio, and now
> with the phone outside of network, we have no way audio. Before
> discussing NAT (which is enabled on the peer), and port forwarding
> (which is setup on the remote location), I would like to make sure I
> fully understand all the sip.conf settings. We are using Asterisk
> realtime via sip_buddies, and the fields in question are:
> 
> (Enclosed in brackets are an example value for the setting)
> 
> * host (dynamic): No problem here. Just wanted to mention that it's
> set as such
> * nat (yes): No problem here either
> * defaultuser (1...@example.com): Does the "@example.com" have to
> point to the UA (i.e., (1003@ua-public-ip), or is it just a name type
> field?
> * fullcontact: What to put here for a UA that is running at a remote
> location with dynamic external IP?
> * ipaddr (ua-public-ip): I did try setting it to the public ip of the
> UA, but is that really practical?
> What if I don't know where the initial registration request is coming
> from? I am guessing "host=dynamic" takes care of that.
> * defaultip??
> * dynamic: Should this be set to yes, or is "host=dynamic" sufficient?
> 
> The phone registers fine, and terminates a call through our providers.
> Just no audio both ways, which would suggest something more that an
> RTP issue which should at least have one way outgoing audio.
> 
> Things that have been attempted:
> * Port forwarding to the phone
> * Changing defaultuser to 1003@ua-public-ip. This made our OpenSIPS
> sip proxy through a fit.
> 
> Things I will attempt today:
> Calling the UA extension from an extension here
> SIP trace
> 
> Your help is greatly appreciated!!!
> 
> Nick.
> 

Hi

Is your directmedia/canreinvite (depending on version) set to no?

Regards

Ish

-- 
Ishfaq Malik 
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552


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Re: [asterisk-users] Dialing out and recording

[Henrik Westerberg]

Yes I should really upgrade, just have to make sure that asterisk-java
will work properly with 1.8

/H








Den 2013-01-02 22:25 skrev Danny Nicholas :

>1.6.2 is a "deader soldier" than 1.4.X.
>
>-Original Message-
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
>Westerberg
>Sent: Wednesday, January 02, 2013 3:20 PM
>To: asterisk-users@lists.digium.com
>Subject: Re: [asterisk-users] Dialing out and recording
>
>#2 works for me on Asterisk 1.8.12 when setting the header like this:
>
>exten => _S,n,SipSetHeader("Diversion: " ${CALLERID(rdnis)})
>
>I haven't been able to make it work on 1.6 yet though, has anyone else?
>
>
>/Henrik
>
>
>>
>>
>>
>> 
>>
>>From: asterisk-users-boun...@lists.digium.com
>>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
>>Sent: Wednesday, January 02, 2013 9:32 AM
>>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>>Subject: Re: [asterisk-users] Dialing out and recording
>>
>> 
>>
>>I have the same requirement, but it's important that the caller ID
>>information from the original caller is presented to the destination
>>and we announce the call before the "transfer" is complete. The carrier
>>requires a diversion header if the ANI is not one of "our" DIDs. Does
>>someone have experience with this working?
>>
>>--
>>
>>Two suggestions for you, Don.  #1 if the Dial is "Private" the
>>"announcement" is taken care of. #2 I'm supposing that you could do a
>>"SIP Header" command before the Dial to resolve the diversion header
>>issue.
>>
>>-- next part -- An HTML attachment was
>>scrubbed...
>>URL: 
>>>459
>>43b1f/attachment-0001.htm>
>>
>>--
>
>
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