Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2

[unserossi]

Please note that I don't claim myself a guru, just happened to be working with 
Asterisk for some good number of years, so probably know some stuff better than 
others.
As for the number of lines, 1800 lines will come down to 1000 lines using AEL 
but not the opposite.
When I'll be back home, hopefully tomorrow, after a beautiful tour (my first) 
of New York city, I'll start writing some blogs on AEL. I guess an IVR example 
could be a good point to start, as it is enough complicated in itself.



Zeeshan A Zakaria
--

Sounds great. A confbridge example would be very welcome to me (just to 
contribute a personal wish) :-)
Enjoy the rest of your trip.

Oliver

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[asterisk-users] CDR: MySQL query

[RSCL Mumbai]

Hi,

Can someone help me formulate MySQL Query(s) which will help me extract the
following details for a given DID (date range can be excluded for
simplicity).

Date-Time
DNID (I am recording this is `userfield`)
CLID
time-1 (when call was received)
time-2 (when call was answered by agent)
time-3 (when call was hung-up)

My Call flow is as follows:
- Caller dials a DNID
- Call enters queue
- Call rings in round-robin format to all logged in agents
- Agent answers call
- Both parties hand-up



Any help with MySQL queries or pointers are deeply appreciated.

Thx
Sans
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Re: [asterisk-users] outboundproxy timeout or qualify

[Abeed Saleh]

Hi Paul,

Thank you for your reply. Unfortunately the timeout parameter will not do
the job for me. I need something equivalent to "qualify" to monitor the
outboundproxy.

Best

On Tue, Aug 3, 2010 at 7:25 PM, Paul Belanger
wrote:

> On Tue, Aug 3, 2010 at 9:58 PM, Abeed Saleh  wrote:
> > Let's say I call by SIP/trunk1/number and the proxy server is down, is
> there
> > a way to get CHANUNAVAIL?
> >
> *CLI> core show application Dial
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
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Re: [asterisk-users] outboundproxy timeout or qualify

[Paul Belanger]

On Tue, Aug 3, 2010 at 9:58 PM, Abeed Saleh  wrote:
> Let's say I call by SIP/trunk1/number and the proxy server is down, is there
> a way to get CHANUNAVAIL?
>
*CLI> core show application Dial

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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Re: [asterisk-users] outboundproxy timeout or qualify

[Abeed Saleh]

Any one can help please?

I go over the problem again.
Let's say I have the following peers

[trunk1]
host=sip.provider.com
outboundproxy=sbc1.probider.com
type=peer

[trunk2]
host=sip.provider.com
outboundproxy=sbc2.provider.com
type=peer

Let's say I call by SIP/trunk1/number and the proxy server is down, is there
a way to get CHANUNAVAIL?

Any suggestions?

Thank you



On Tue, Aug 3, 2010 at 10:39 AM, Abeed Saleh  wrote:

> Hi All,
>
> I'm connecting to my carrier which requires setting of outboundproxy. There
> has been few cases where the proxy server failed due to network issues and
> required us to use a secondary one. Is there a timeout or qualify setting
> for outboundproxy setting in sip.conf?
>
> I do appreciate if anyone can help please.
>
> Thank you
>
> -Abeed
>
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Re: [asterisk-users] Using SIP to dial extension that will give anoutside line

[Warren Selby]

On Tue, Aug 3, 2010 at 4:49 PM,  wrote:

>  Unfortunately it is only the Iwatsu IP phones that grab the open line @
> 3001 currently, the softphones do not.  I might try programming the
> extension and see if I can get a response that way.
>
> Mostly what I am seeing is 
>

Assuming you have a peer defined in sip.conf as Iwatsu, you could try this:

exten => 1234,1,Verbose(Making outbound test call over Iwatsu trunk)
exten => 1234,n,Dial(SIP/Iwatsu/3001,D(ww${DIALNUMBER}))

where ${DIALNUMBER} is the number you'd like to dial.  For testing I would
set this to a static number (your cell phone or whatever).  Once you have
that working, you would implement it for real in a fashion similar to this:

[outgoing-dial]
exten => _X.,1,Verbose(Making outbound call over Iwatsu trunk)
exten => _X.,n,Dial(SIP/Iwatsu/3001,D(ww${EXTEN}))

and then make the outgoing-dial context available from whatever context your
phone defaults to.

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] Using SIP to dial extension that will give anoutside line

[Jeremy.Hellstrom]

-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of Carlos Chavez
Sent: Tue 8/3/2010 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Using SIP to dial extension that will give 
anoutside line
 
On Tue, 2010-08-03 at 16:04 -0500, Danny Nicholas wrote:
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> jeremy.hellst...@synovate.com
> Subject: [asterisk-users] Using SIP to dial extension that will give
> anoutside line
> 
> 
>  
> 
> You could try this:
> 
>  
> 
> ; use lwatsu line
> 
> Exten => 1234,1,dial(SIP/3001ww5551212)
> 
>  
> 
> If dialing extension SIP/3001 from asterisk connects to the lwatsu
> with an open line, the ww5551212 will wait 1 second, the dial on using
> the lwatsu.
> 
>   Actually, you nee to dial like this:
>
>exten => 1234,1,Dial(SIP/lwatsu_sip/${NUMBER})
>
>lwatsu_sip must be a defined peer in your sip.conf and ${NUMBER} would
>be the number you wish to dial through that peer.  If you need to send
>the DTMF after the call is connected you can use the D option in the
>dial command.  It is up to the PBX to interpret the number you sent
>using its internal dialplan.
>
.
>-- 
>Telecomunicaciones Abiertas de México S.A. de C.V.
>Carlos Chávez Prats
>Director de Tecnología
>+52-55-91169161 ext 2001


Thanks all, 
Unfortunately it is only the Iwatsu IP phones that grab the open line @ 3001 
currently, the softphones do not.  I might try programming the extension and 
see if I can get a response that way.

Mostly what I am seeing is 

*CLI>   == Using SIP RTP CoS mark 5
-- Executing [96046642...@phones:1] Dial("SIP/testphone1-0053", 
"SIP/6046642400") in new stack
  == Using SIP RTP CoS mark 5
[Aug  3 14:41:02] WARNING[1948]: chan_sip.c:5340 create_addr: No such host: 
6046642400
[Aug  3 14:41:02] WARNING[1948]: app_dial.c:1747 dial_exec_full: Unable to 
create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [96046642...@phones:2] Congestion("SIP/testphone1-0053", 
"") in new stack
  == Spawn extension (phones, 96046642400, 2) exited non-zero on 
'SIP/testphone1-0053'

or

*CLI>   == Using SIP RTP CoS mark 5
-- Executing [96046642...@phones:1] Dial("SIP/testphone1-0057", 
"SIP/Iwatsu/6046642400") in new stack
  == Using SIP RTP CoS mark 5
-- Called Iwatsu/6046642400
[Aug  3 14:47:36] WARNING[3239]: chan_sip.c:17865 handle_response_invite: 
Received response: "Forbidden" from '"TestPhone1" 
;tag=as60718fca'
-- SIP/Iwatsu-0058 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [96046642...@phones:2] Congestion("SIP/testphone1-0057", 
"") in new stack
  == Spawn extension (phones, 96046642400, 2) exited non-zero on 
'SIP/testphone1-0057'


Dependent on defining Iwatsu as a friend in the latter or as a variable in the 
former.  By the way Exten => 1234,1,dial(SIP/3001ww5551212) had asterisk return 
No such host: 3001ww5551212
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Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2

[Joel Maslak]

On Tue, Aug 3, 2010 at 1:30 PM, Mark G. Thomas  wrote:


> I didn't know there was a "U" option. I don't see any mention of it
> on the voip-info.org wiki or other Dial() documentation, but didn't
> check for new options in the built in documentation until just now.
>


I updated the dial documentation on voip-info.org - but I'm sure I didn't do
it perfectly.  I also re-ordered the options by ASCII sort order, rather
than the random order (I think) that was there before.
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Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

[bruce bruce]

Hi Mike,

I am putting the phones on AC Adapter now as I am suspecting the Linksys POE
switch. Once that test is done and if problem still presists, I will be
enabling DHCPMasq and also set the SIP registration time to 1 second on the
phone UI.

-Bruce

On Tue, Aug 3, 2010 at 1:13 PM, Mike  wrote:

>  Hi Bruce,
>
>
>
> Did you ever get a working solution and confirm the underlying issue ? I am
> having the same issue on a set of phones, my next step is replacing the
> router, but I was wondering if you found something else.
>
>
>
> Regards,
>
>
>
> Mike
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
> *Sent:* Thursday, July 29, 2010 22:36
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Aastra phones occasionally show "No Service" -
> Is there any network setting I can tamper to facilitate a quick DHCP renewal
> on the Aastra phones?
>
>
>
> Hi Everyone,
>
>
>
> I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The
> phones occasionally go into "No Service" mode. The POE switch doesn't seem
> to be the problem as it's tested fine. I think the router sometimes gives up
> and comes back quickly. Or something of that nature. However, the
> connections are maintained if a call is going on because there are peer to
> peer connections between the phones in a network. Anyhow, if the phones are
> restarted they work fine.
>
>
>
> So, I was looking around the Aastra Admin UI to find any timer to lower it
> to 1 second to check and make sure the device always has an ip but I can't
> seem to find anything other than LLDP which is set at 30 and I don't think
> that will be of any help.
>
>
>
> I did a test where I would disconnect the router from the switch and after
> a while phones go into "No Service" but if I plug it back into the switch
> the phones do not come back right away. Maybe something should be dialed on
> the phone or wait long time or restart it to work again.
>
>
>
> Any work around?
>
>
>
> Thanks a lot
>
> --
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Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2

[Zeeshan Zakaria]

Please note that I don't claim myself a guru, just happened to be working
with Asterisk for some good number of years, so probably know some stuff
better than others.

As for the number of lines, 1800 lines will come down to 1000 lines using
AEL but not the opposite.

When I'll be back home, hopefully tomorrow, after a beautiful tour (my
first) of New York city, I'll start writing some blogs on AEL. I guess an
IVR example could be a good point to start, as it is enough complicated in
itself.


Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-08-03 4:49 PM, "Danny Nicholas"  wrote:

  *>From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *unsero...@aol.com
>*Subject:* Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2





>AEL is very simple and the instructions on voip-info.org are enough to
learn it. In fact I can...

I’ve only been with Asterisk since 1.4.18; the “programmer” in me still
finds it simpler to do a 1000 line extensions.conf vs an 1800 line readable
AEL but the examples on www.ilovetovoip.com are going to bring me around
more quickly than the voip-info.org stuff. You write some pretty good stuff,
Z.

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Re: [asterisk-users] FYI: Seen the 2600Hz announcement?

[Randy R]

We should be hearing more on this from Darren either this Friday or next on VUC.

http://vuc.me

/r

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Re: [asterisk-users] Using SIP to dial extension that will give anoutside line

[Carlos Chavez]

On Tue, 2010-08-03 at 16:04 -0500, Danny Nicholas wrote:
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> jeremy.hellst...@synovate.com
> Subject: [asterisk-users] Using SIP to dial extension that will give
> anoutside line
> 
> 
>  
> 
> You could try this:
> 
>  
> 
> ; use lwatsu line
> 
> Exten => 1234,1,dial(SIP/3001ww5551212)
> 
>  
> 
> If dialing extension SIP/3001 from asterisk connects to the lwatsu
> with an open line, the ww5551212 will wait 1 second, the dial on using
> the lwatsu.
> 
Actually, you nee to dial like this:

exten => 1234,1,Dial(SIP/lwatsu_sip/${NUMBER})

lwatsu_sip must be a defined peer in your sip.conf and ${NUMBER} would
be the number you wish to dial through that peer.  If you need to send
the DTMF after the call is connected you can use the D option in the
dial command.  It is up to the PBX to interpret the number you sent
using its internal dialplan.


-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Using SIP to dial extension that will give anoutside line

[Danny Nicholas]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
jeremy.hellst...@synovate.com
Subject: [asterisk-users] Using SIP to dial extension that will give
anoutside line

 

You could try this:

 

; use lwatsu line

Exten => 1234,1,dial(SIP/3001ww5551212)

 

If dialing extension SIP/3001 from asterisk connects to the lwatsu with an
open line, the ww5551212 will wait 1 second, the dial on using the lwatsu.

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[asterisk-users] Using SIP to dial extension that will give an outside line

[Jeremy.Hellstrom]

I am trying to add an Asterisk box to an Iwatsu ECS (Software Version
7.0 R.01) hopefully without using a physical T1/E1 card.  Internally the
SIP works fine, it is dialling an outside line that is giving me
difficulties.  One way that I think it might be possible is for an
outbound call to connect to extenstion 3001, which is one of 36 PRI
trunks available to the Iwatsu system.  Dialling ext 3001 on an Iwatsu
immediately gives an Iwatsu phone an open outbound line.

 

I cannot figure out a way to define that extension in the dial plan when
I enter the Iwatsu as a channel.  Am I totally barking up the wrong tree
with this method?

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Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2

[Danny Nicholas]

>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
unsero...@aol.com
>Subject: Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2

 

>AEL is very simple and the instructions on voip-info.org are enough to
learn it. In fact I can't understand how can one write complex dial plans
not using AEL, you simply >can't do it using standard format used in
extensions.conf.

>As for the tutorials, there is no specific website for them as per my
knowledge, but I can certainly try to write some basic ones on my blog to
help those whom it might >help. Many examples in my blogs are already in
AEL, from real life scenarios.

>Zeeshan A Zakaria

 

 

>I'd love to see your real life examples as from what i found is that most
of the information on voip-info seems to be >outdated and new features/
applications are not documented or examples of how to use them and get all
possible benefits >out of it are missing.

>Especially for beginners not working with Asterisk since version 1.0 or so
it is really hard to get into it without >having some complex dialplan
examples to learn from.

>Oliver

 

I've only been with Asterisk since 1.4.18; the "programmer" in me still
finds it simpler to do a 1000 line extensions.conf vs an 1800 line readable
AEL but the examples on www.ilovetovoip.com 
are going to bring me around more quickly than the voip-info.org stuff. You
write some pretty good stuff, Z.

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Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2

[unserossi]

AEL is very simple and the instructions on voip-info.org are enough to learn 
it. In fact I can't understand how can one write complex dial plans not using 
AEL, you simply can't do it using standard format used in extensions.conf.
As for the tutorials, there is no specific website for them as per my 
knowledge, but I can certainly try to write some basic ones on my blog to help 
those whom it might help. Many examples in my blogs are already in AEL, from 
real life scenarios.

Zeeshan A Zakaria
--

I'd love to see your real life examples as from what i found is that most of 
the information on voip-info seems to be outdated and new features/ 
applications are not documented or examples of how to use them and get all 
possible benefits out of it are missing.

Especially for beginners not working with Asterisk since version 1.0 or so it 
is really hard to get into it without having some complex dialplan examples to 
learn from.

Oliver

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Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2

[Zeeshan Zakaria]

AEL is very simple and the instructions on voip-info.org are enough to learn
it. In fact I can't understand how can one write complex dial plans not
using AEL, you simply can't do it using standard format used in
extensions.conf.

As for the tutorials, there is no specific website for them as per my
knowledge, but I can certainly try to write some basic ones on my blog to
help those whom it might help. Many examples in my blogs are already in AEL,
from real life scenarios.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-08-03 3:36 PM, "Mark G. Thomas"  wrote:

Hi,


On Tue, Aug 03, 2010 at 01:49:11PM -0500, Tilghman Lesher wrote:
> On Tuesday 03 August 2010 13:19:...
Thank you!

I didn't know there was a "U" option. I don't see any mention of it
on the voip-info.org wiki or other Dial() documentation, but didn't
check for new options in the built in documentation until just now.

Mark


-- 
Mark G. Thomas (m...@misty.com)

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Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2

[Mark G. Thomas]

Hi,

On Tue, Aug 03, 2010 at 01:49:11PM -0500, Tilghman Lesher wrote:
> On Tuesday 03 August 2010 13:19:11 Mark G. Thomas wrote:
> > I can't figure out what syntax to use with the Dial() "M" parameter
> > for the AEL parser to interpret properly.  Creating an AEL
> > macro named "macro-screen()" partly works as a hack, but it must
> > not turn into a gosub properly, so I get warnings about the "return;".
> 
> Is there a reason you don't want to use the 'U' option in Dial?  It was
> created specifically for this purpose.

Thank you!

I didn't know there was a "U" option. I don't see any mention of it
on the voip-info.org wiki or other Dial() documentation, but didn't
check for new options in the built in documentation until just now.

Mark

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Re: [asterisk-users] FYI: Seen the 2600Hz announcement?

[dotnetdub]

On 3 August 2010 19:54, Paul Belanger  wrote:

> On Tue, Aug 3, 2010 at 2:26 PM, Duncan Turnbull 
> wrote:
> > FreePBX is still the same, V3 is still the same, this is a fork from some
> guys who had got involved (or maybe paid some money)
> >
> That is how I read the announcement.



from freepbx.com:

There have been some inquiries over the last few weeks along with some
additional news that has come out concerning the direction of FreePBX v3 and
the lack of activity in SVN.

Rest assured that the intensive work that has gone into this project is live
and well but will be continuing under a new project name.

When we set off with the development of v3 we wanted to rewrite a system
from ground up and that is exactly how the project was executed. Darren
(pyte) was and has been the chief architect of v3 from the beginning. In an
attempt to make sure v3 could flourish with maximum creativity and not
necessarily be hampered with any "baggage" from v2, the projects were run
fairly independently though they shared this same website.

The fact that they were both under the same "FreePBX" umbrella started to
create a bit of confusion in places like the Forums and IRC channel. There
was also often concern that v2 development, which supports an installed base
of probably 500K systems, would be slowed because of v3. Given that v3 has
been a rewrite from ground up there was also concern by some when its
feature set in the early "betas" was a subset in many ways of what AMP 1.0
had 5.5 years ago.

It became clear that the best thing that could be done for both projects was
to spin v3 off into its own identitty thus allowing both FreePBX and the new
rewrite to flourish and serve the community in the best possible way. The
new project, still run by Darren, is named 2600hz
Project and
will be the new home to allow v3 to flourish, while FreePBX (v2) continues
to evolve and serve the large installed base that it enjoys today.

We will be cleaning up the current web site over the next week to reflect
these directions and are excited as always for all the great work that is
being done in this place and by all the contributors that help us make such
a great project!

*Philippe* on behalf of the FreePBX Team
http://freepbx.com/news/2010-08-03/v3-spun-off-to-give-it-full-independence
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Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2

[Tilghman Lesher]

On Tuesday 03 August 2010 13:19:11 Mark G. Thomas wrote:
> I can't figure out what syntax to use with the Dial() "M" parameter
> for the AEL parser to interpret properly.  Creating an AEL
> macro named "macro-screen()" partly works as a hack, but it must
> not turn into a gosub properly, so I get warnings about the "return;".

Is there a reason you don't want to use the 'U' option in Dial?  It was
created specifically for this purpose.

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Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2

[Danny Nicholas]

>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark G. Thomas
>Subject: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2

>Hi,

>I can't figure out what syntax to use with the Dial() "M" parameter
>for the AEL parser to interpret properly.  Creating an AEL 
>macro named "macro-screen()" partly works as a hack, but it must
>not turn into a gosub properly, so I get warnings about the "return;".

>  Dial(...,tgM(&screen)) with the ael macro named "screen" does not work
>  Dial(...,tgM&screen) with the ael macro named "screen" does not work
>  Dial(...,tgM(screen)) with the ael macro named "screen" does not work
>  Dial(...,tgM(screen)) with the ael macro named "macro-screen" partly
works

>Is a more correct or otherwise better way to do this in AEL? Is there
>some other solution? The other followme examples I've found all have
>different behaviors than I want. I'm not looking for the caller to
>be prompted for their name or anything, and I don't want the followme
>connect to happen unless the cellphone user hits a "1" to accept the 
>call, or other key to ditch the call, otherwise cellphone voicemail 
>gets the call.

>Mark

Personally, I haven't gotten too involved in AEL because it's cumbersome (to
me) to keep up across platforms (CENTOS/Suse/etc) but I have found that AEL
is "additive"; you can write "troublesome" code using the old
extensions.conf nomenclature so you don't have to worry about how AEL
creates it.  Not the cleanest or best solution, but it should be workable.

P.S.  All of you who are going to flame me, how about posting some good AEL
tutorial links?


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Re: [asterisk-users] FYI: Seen the 2600Hz announcement?

[Paul Belanger]

On Tue, Aug 3, 2010 at 2:26 PM, Duncan Turnbull  wrote:
> FreePBX is still the same, V3 is still the same, this is a fork from some 
> guys who had got involved (or maybe paid some money)
>
That is how I read the announcement.

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Polybeacon | Consultant
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blog.polybeacon.com

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Re: [asterisk-users] FYI: Seen the 2600Hz announcement?

[Duncan Turnbull]

No its a split

FreePBX is still the same, V3 is still the same, this is a fork from some guys 
who had got involved (or maybe paid some money)

Cheers Duncan

On 4/08/2010, at 2:56 AM, Tzafrir Cohen wrote:

> On Tue, Aug 03, 2010 at 02:28:15PM +0100, Alan Lord (News) wrote:
>> http://gigaom.com/2010/08/03/2600hz-project/
> 
> So practically FreePBX V3 was renmed 2600Hz / BlueBox ?
> 
> -- 
>   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] Dial() M parameter in 1.6.2.11-rc2

[Mark G. Thomas]

Hi,

I can't figure out what syntax to use with the Dial() "M" parameter
for the AEL parser to interpret properly.  Creating an AEL 
macro named "macro-screen()" partly works as a hack, but it must
not turn into a gosub properly, so I get warnings about the "return;".

  Dial(...,tgM(&screen)) with the ael macro named "screen" does not work
  Dial(...,tgM&screen) with the ael macro named "screen" does not work
  Dial(...,tgM(screen)) with the ael macro named "screen" does not work
  Dial(...,tgM(screen)) with the ael macro named "macro-screen" partly works

Is a more correct or otherwise better way to do this in AEL? Is there
some other solution? The other followme examples I've found all have
different behaviors than I want. I'm not looking for the caller to
be prompted for their name or anything, and I don't want the followme
connect to happen unless the cellphone user hits a "1" to accept the 
call, or other key to ditch the call, otherwise cellphone voicemail 
gets the call.

Mark

context inbound {
...
211234 => Dial(SIP/1...@cme&local/1...@internals,18,rt); // screen these
...
};

context internals {
102 => {
Dial(${CELLPHONE},30,tgM(screen)); // cellphone user gets prompted
jump s...@general-menu; // jump to IVR menu if call not accepted
};
};

// play message to cellphone before connecting inbound call
// http://www.voip-info.org/wiki/view/Asterisk+tips+findme
// http://lists.digium.com/pipermail/asterisk-dev/2005-June/013598.html
//
macro macro-screen() {// hack
Wait(0.5);
Read(ACCEPT,followme/options,1,,1,20);
if( "${ACCEPT}" = "1" ) {
Background(connecting);
} else {
Set(MACRO_RESULT=CONTINUE);
};
return;   // I get AEL complaints regardless of whether this is here or not.
};


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Re: [asterisk-users] fail2ban does not work for my asterisk installation

[mosbah abdelkader]

Thank you doctor whom,


It is working for me now.
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[asterisk-users] outboundproxy timeout or qualify

[Abeed Saleh]

Hi All,

I'm connecting to my carrier which requires setting of outboundproxy. There
has been few cases where the proxy server failed due to network issues and
required us to use a secondary one. Is there a timeout or qualify setting
for outboundproxy setting in sip.conf?

I do appreciate if anyone can help please.

Thank you

-Abeed
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Re: [asterisk-users] real-time queue problems

[Cristian Dimache]

Hello Al,

On 03.08.2010 20:06, Alvaro Ramirez wrote:
> Any body experiencing incoming calls into a sip-agent which is 
> already registered into a real-time queue with Asterisk, having an 
> active call?
> We are using Asterisk Version 1.4.21.2

 Yes - see https://issues.asterisk.org/view.php?id=17358 and 
https://issues.asterisk.org/view.php?id=17773
 No idea about a solution yet.

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[asterisk-users] ConfBridge

[iscario]

Hi,
I have a question about the use of Confbridge.
In my dialplan, I allow a caller to reserve a conference room by choosing a room
number and a password. Thus, any other caller who would like to join the same
room can do it only if he knows the password. It works fine, but I need to free
the reservation at a time, otherwise nobody else can reserve the same room for
another use. I store the variables in the Asterisk database (one to tell if the
room is busy, one to save the password).
Till now, I launched in background a script (thanks to the AGI) which erases the
database variables used for the reservation after a defined time. Not ideal,
because I don’t even know if there is some people in the room after the defined
time…. So I would like to know if there is a better way to do this.
Is it possible to know (eg via the AMI) if somebody is in a Confbridge room
(knowing the room number of course) ?
Or is it possible to tell directly to Confbridge to close the conference room
after a while ? (but I would still have to erase my variables)

Thanks!



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Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

[Mike]

Hi Bruce,

 

Did you ever get a working solution and confirm the underlying issue ? I am
having the same issue on a set of phones, my next step is replacing the
router, but I was wondering if you found something else.

 

Regards,

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Thursday, July 29, 2010 22:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Aastra phones occasionally show "No Service" - Is
there any network setting I can tamper to facilitate a quick DHCP renewal on
the Aastra phones?

 

Hi Everyone,

 

I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The phones
occasionally go into "No Service" mode. The POE switch doesn't seem to be
the problem as it's tested fine. I think the router sometimes gives up and
comes back quickly. Or something of that nature. However, the connections
are maintained if a call is going on because there are peer to peer
connections between the phones in a network. Anyhow, if the phones are
restarted they work fine.

 

So, I was looking around the Aastra Admin UI to find any timer to lower it
to 1 second to check and make sure the device always has an ip but I can't
seem to find anything other than LLDP which is set at 30 and I don't think
that will be of any help. 

 

I did a test where I would disconnect the router from the switch and after a
while phones go into "No Service" but if I plug it back into the switch the
phones do not come back right away. Maybe something should be dialed on the
phone or wait long time or restart it to work again.

 

Any work around?

 

Thanks a lot

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Re: [asterisk-users] Garbled messages - format_wav_gsm.c:414 wav_read:Short read (60) (No such file or directory)!

[Danny Nicholas]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bobby Larson
Subject: [asterisk-users] Garbled messages - format_wav_gsm.c:414
wav_read:Short read (60) (No such file or directory)!

 

The problem is that voice mail message get played back garbled.
Occasionally, I can make out moments of a voice or another sound that may be
in the actual message, though it's far too diluted with garbling and chirps
to detect any words or phrases.  When running asterisk -r, I will get a
message on the console after the system completes playback of the file:

 

format_wav_gsm.c:414 wav_read: Short read (60) (No such file or directory)!

 

I am running under a linux virtual server, and here is the output of "core
show version":

 

Asterisk SVN-trunk-r161350 built by root @ ast01 on a x86_64 running Linux
on 2008-12-05 17:31:32 UTC

 

Though this problem is an old one, it has gradually gotten much worse over
time.  Originally, I just thought it was the result of excessive network
traffic, but I'm starting to really think otherwise.

 

My current workaround is to manually copy over the wav files to a local
computer for listening.  This says to me that recording is working fine,
it's the playback that has problems.

 

In your /etc/asterisk/voicemail.conf (or equivalent), I assume you have this
line:

Format = wav

Or 

Format = wav|gsm|wav49

 

The wav49 (.WAV) format is "friendlier" (in this poster's opinion) for
playback.

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[asterisk-users] real-time queue problems

[Alvaro Ramirez]

Any body experiencing incoming calls into a sip-agent which is
already registered into a real-time queue with Asterisk, having an active
call?
We are using Asterisk Version 1.4.21.2
Al
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[asterisk-users] Garbled messages - format_wav_gsm.c:414 wav_read: Short read (60) (No such file or directory)!

[Bobby Larson]

I am having a problem with asterisk voice mail messages that seems to be
intermittent.  Though the problem occurs most of the time, on rare occasions
it will work fine - rare enough that I can't pin down what it is that works.

The problem is that voice mail message get played back garbled.
 Occasionally, I can make out moments of a voice or another sound that may
be in the actual message, though it's far too diluted with garbling and
chirps to detect any words or phrases.  When running asterisk -r, I will get
a message on the console after the system completes playback of the file:

format_wav_gsm.c:414 wav_read: Short read (60) (No such file or directory)!

I am running under a linux virtual server, and here is the output of "core
show version":

Asterisk SVN-trunk-r161350 built by root @ ast01 on a x86_64 running Linux
on 2008-12-05 17:31:32 UTC


I am attempting to listen to the messages from a SIP phone and a SIP
softphone, and it doesn't make any difference where I am or what I use.

Though this problem is an old one, it has gradually gotten much worse over
time.  Originally, I just thought it was the result of excessive network
traffic, but I'm starting to really think otherwise.

My current workaround is to manually copy over the wav files to a local
computer for listening.  This says to me that recording is working fine,
it's the playback that has problems.

Thanks for any help.

:Bobby
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Re: [asterisk-users] What do you use for Invoicing?

[Jeff LaCoursiere]

On Tue, 2010-08-03 at 09:17 -0400, bruce bruce wrote:
> I agree but the mentioned software is not opensource. 
> My conditions clearly included opensource.
> 

No, your "prefer" listed "opensource".  If you had said "requirement" I
wouldn't have suggested it.

j

> On Tue, Aug 3, 2010 at 12:35 AM, Nick Brown  wrote:
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> bruce bruce
> Sent: Tuesday, 3 August 2010 1:58 PM
> To: j...@sunfone.com; Asterisk Users Mailing List -
> Non-Commercial Discussion
> 
> 
> Subject: Re: [asterisk-users] What do you use for Invoicing?
>  
> 
> Maybe good but the first look brought me to a Pay version.
> Doesn't satisfy the opensource condition.
> 
>  
> 
> 
> thanks,
> 
>  
> 
> Open Source software does not necessarily mean free software. 
> 
>  
> 
> Nick.
> 
>  
> 
>  
> 
> 
> 
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Re: [asterisk-users] What do you use for Invoicing?

[Leif Madsen]

On 10-08-02 02:26 PM, bruce bruce wrote:
> Hi Everyone,
>
> Sorry, if it's not directly related to Asterisk. Some of people on this
> list might have PBX deployed for their clients. What software do you use
> to invoice them so the invoice looks like a proper telecom invoice maybe?
>
> Prefer:
> -opensource with Windows binary available.
> -able to create .pdf invoices rather than printable ones.

This may not meet all specifications (as I haven't played with it yet), but I 
saw the full blown version in a demo a couple of months ago and it was 
definitely impressive. It had a lot of metrics and data for billing, and did 
everything I could think of. I just don't have any customers who have required 
such a service yet.

They also have a free service, but again, I haven't tried it, but it might be 
something to check out:

http://dthfreecallrating.com/

The pay service is here:

http://www.dthvoipbilling.com/

Leif Madsen.

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Re: [asterisk-users] FYI: Seen the 2600Hz announcement?

[Dean Collins]

wow which idiot decided to associate this project with call phreaking.

really dumb move.

Shame this is where AMP has ended up based on where it started from.





Regards,

Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357   New York
+61-2-9016-5642   (Sydney in-dial).
+44-20-3129-6001 (London in-dial).

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
> Sent: Tuesday, 3 August 2010 10:57 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] FYI: Seen the 2600Hz announcement?
> 
> On Tue, Aug 03, 2010 at 02:28:15PM +0100, Alan Lord (News) wrote:
> > http://gigaom.com/2010/08/03/2600hz-project/
> 
> So practically FreePBX V3 was renmed 2600Hz / BlueBox ?
> 
> --
>Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
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> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
> 
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Re: [asterisk-users] asterisk-users Digest, Vol 73, Issue 5

[Nasir Javaid]

> > Question 1 :
> > [Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer -
> > audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)
> > why is combined alaw|g726 and not g726|alaw (reverse) ??
>
> Guess: Here the order presented has no meaning for the order of codec
> negotiation.
>
> > Question 2 :
> > why do I see on my Grandstream phone that the codec being used is alaw in
> > stead of g726 ??
>
> Because that is what the phone and Asterisk have negotiated. ;-)
>
> > Question 3 :
> > How can I get g726 as first preferred codec ??
>
> Which Asterisk version are you using?
>
> * check if you have disallow/allow settings in the [general] section of
> sip.conf. Depending on your Asterisk version only the order in [general]
> would be respected, but not the order in the individual sip peer/user
> definition
>
> * look at the variable SIP_CODEC for the inbound (first) call leg, and in
> Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND
>
> * many Asterisk operators have applied the third party "codec negotiation
> patch"
>
> Philipp
>
>
>
>
> --
>
> Message: 15
> Date: Tue, 3 Aug 2010 07:26:41 -0400
> From: C F 
> Subject: Re: [asterisk-users] Caller ID issue
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID:
>
> Content-Type: text/plain; charset=ISO-8859-1
>
> In most cases wait(.5) will do. I would not recommend using
> answer(2000) as that answers the channel, which means you start
> getting billed.
>
> On 8/2/10, Peder  wrote:
> >> I am using T1's and didn't think the spill would take that long.
> >
> >> PRI no, E&M yes.
> >
> > Some PRI take that long too because the telco sends the name in a
> followup
> > message, not in the initial call setup.
> >
> >
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
>
> Message: 16
> Date: Tue, 03 Aug 2010 07:51:51 -0400
> From: John Novack 
> Subject: Re: [asterisk-users] chinaroby fxo card - never heard of them
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID: <4c5802d7.3080...@stromberg-carlson.org>
> Content-Type: text/plain; charset="iso-8859-1"
>
> They seem to have taken over manufacture of cards Digium has discontinued.
> I have used several of the TE110 card with success and they are identical
>
> John Novack
>
>
> asteriskguru asteriskguru wrote:
> > hi,
> > I am using this card and IP phone about 6 months. There is no issues
> > at all.
> >
> > Installation procedures are same as Digium  analog card.
> >
> > Hope it helps,
> > Ashik
> >
> > On Tue, Aug 3, 2010 at 6:28 AM, Landy Landy  > <mailto:landysacco...@yahoo.com>> wrote:
> >
> > Hello.
> >
> > I'm looking to buy a FXO card to do some testing with two phone
> > lines I have at home and was looking in ebay some and found some
> > cheap ones but, the I've never heard of the brand or manufacturer:
> > chinaroby. They run for about $99 plus shipping. Have any one used
> > these? or please recommend one... Money IS an issue.
> >
> > Thanks.
> >
> >
> >
> >
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> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
> --
>
> Dog is my Co-pilot
>
> -- next part --
> An HTML attachment was scrubbed...
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>
> --
>
> Message: 17
> Date: Tue, 3 Aug 2010 09:11:07 -0400
> From: Zeeshan Zakaria 
> Subject: Re: [asterisk-users] RTP stream no

Re: [asterisk-users] Fax/Modem, Asterisk, Channel Banks

[Doug Lytle]

Joel Maslak wrote:
>
> So...will this work?


It will work very well, I have two installations with ADIT600s

Doug


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Re: [asterisk-users] Fax/Modem, Asterisk, Channel Banks

[Tim Nelson]

- "Joel Maslak"  wrote: 
> I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm 
> needing a solution for fax machines that works as well as a POTS line from my 
> carrier. If the POTS line is the solution, I'll keep it, but I'd rather move 
> away from that. 
> 
> Here's what I'm thinking...will it work? 
> 
> I would use a dual-port Digium T1 card. In one port, I'd terminate a telco 
> PRI T1. In the other port, I'd terminate a Rhino channel bank, connected to 
> each of my fax machines (and a stamp machine with an internal modem). 
> 
> What I'm wanting is to be able to send/receive faxes via the telco PRI and 
> the analog fax machines. I also want the stamp machine to work. I don't want 
> this to work 98% as well as the Telco - they truly need to work 100% as well. 
> 
> So...will this work? 
> 


Be absolutely sure you have your Zaptel/DAHDI timing settings correct so all 
ports are sync'ed properly with one timing source. Also, ensure your telco is 
giving you a real TDM circuit/path, not a TDM port that is serviced by a VoIP 
connection. Many companies are doing this now as it cuts their costs 
significantly for service delivery and works fine for voice but you'll have 
problems galore trying to get a modem/fax carrier working reliably. They'll 
usually call this product a 'flex T1' or 'Mega T1' or some other marketing-hype 
type name and give it a lower cost. You don't want to walk away from these 
products, but rather, run as fast as you can. :-) 


I've used the Rhino channel banks extensively and they work very well. In fact, 
a recent installation was specifically for combined voice/fax/modem usage and 
it performed perfectly. The only difference in my case versus yours was that 
the T1 was being provided by the telco, not via a T1 card in another Asterisk 
box. But again, assuming you have your timing setup correctly, this should work 
very well. 

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Re: [asterisk-users] FYI: Seen the 2600Hz announcement?

[C F]

Isn't that trademarked? :P

On Tue, Aug 3, 2010 at 9:28 AM, Alan Lord (News)  wrote:
> http://gigaom.com/2010/08/03/2600hz-project/
>
> --
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Re: [asterisk-users] fail2ban does not work for my asterisk installation

[Lonnie Abelbeck]

Kyle Kienapfel  gmail.com> writes:

> NOTICE.* .*: Registration from '.*' failed for '' -
> ACL error \(permit/deny\)
> 
> 
> I don't see slashes in front of the brackets on what you posted to the
> mailing list. I'm posting my config to see if the mailing list mangles
> it or not.
> 

I think Kyle found the OP's issue.

Additionally, the "- ACL error (permit/deny)" log message was found in
asterisk 1.2.x and no longer seems to occur in the later versions of asterisk.

Lonnie



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Re: [asterisk-users] Fax/Modem, Asterisk, Channel Banks

[Dave Fullerton]

On 08/03/2010 10:48 AM, Joel Maslak wrote:
> I've been replacing an old Toshiba DK switch with an Asterisk solution.  I'm
> needing a solution for fax machines that works as well as a POTS line from
> my carrier.  If the POTS line is the solution, I'll keep it, but I'd rather
> move away from that.
>
> Here's what I'm thinking...will it work?
>
> I would use a dual-port Digium T1 card.  In one port, I'd terminate a telco
> PRI T1.  In the other port, I'd terminate a Rhino channel bank, connected to
> each of my fax machines (and a stamp machine with an internal modem).
>
> What I'm wanting is to be able to send/receive faxes via the telco PRI and
> the analog fax machines.  I also want the stamp machine to work.  I don't
> want this to work 98% as well as the Telco - they truly need to work 100% as
> well.
>
> So...will this work?
>
>

It should. That's the setup I'm using (but with an Adit 600 channel 
bank) and it works perfectly.

-Dave

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Re: [asterisk-users] FYI: Seen the 2600Hz announcement?

[Tzafrir Cohen]

On Tue, Aug 03, 2010 at 02:28:15PM +0100, Alan Lord (News) wrote:
> http://gigaom.com/2010/08/03/2600hz-project/

So practically FreePBX V3 was renmed 2600Hz / BlueBox ?

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[asterisk-users] Fax/Modem, Asterisk, Channel Banks

[Joel Maslak]

I've been replacing an old Toshiba DK switch with an Asterisk solution.  I'm
needing a solution for fax machines that works as well as a POTS line from
my carrier.  If the POTS line is the solution, I'll keep it, but I'd rather
move away from that.

Here's what I'm thinking...will it work?

I would use a dual-port Digium T1 card.  In one port, I'd terminate a telco
PRI T1.  In the other port, I'd terminate a Rhino channel bank, connected to
each of my fax machines (and a stamp machine with an internal modem).

What I'm wanting is to be able to send/receive faxes via the telco PRI and
the analog fax machines.  I also want the stamp machine to work.  I don't
want this to work 98% as well as the Telco - they truly need to work 100% as
well.

So...will this work?
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Re: [asterisk-users] mapping of disconnect reasons

[Tilghman Lesher]

On Tuesday 03 August 2010 06:21:23 Philipp von Klitzing wrote:
> > Is there a way to change the mappings of disconnect reasons to certain
> > SIP messages? E.G. I need to change the mapping for SIP 402 "Payment
> > Required" from 16 (normal termination) like it is in 1.4.24 to 21
> > (call rejected) as defined in RFC 3398.
>
> * if you think the mapping is wrong, then you should open a ticket on the
> Asterisk bug tracker

Actually, much of the mapping is specified by RFC 3398 section 8.2.6.1.  Thus,
if you think the mapping is wrong, you should submit a suggestion for
amendment to the RFC editor.  Only for response codes specified differently
than in this section should you open an issue in the tracker.

-- 
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Digium, Inc. | Senior Software Developer
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Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)

[Philipp von Klitzing]

Hi!

> In the [general] section of sip.conf I have :
> 
> disallow=all
> allow=g726
> allow=alaw
> allow=g729
> allow=gsm

So change the order there and see what happens.

> > * look at the variable SIP_CODEC for the inbound (first) call leg, and
> > in Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND
> 
> When I read the value of this variable just before the Dial()-statement,
> it is empty.

You need to set it, not read it.

Philipp


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Re: [asterisk-users] What do you use for Invoicing?

[Nasir Iqbal]

On Tue, Aug 3, 2010 at 6:24 PM, bruce bruce  wrote:

> Oh, you seem to be right on. It's actually an install of Elastix. I will be
> testing this for sure. Hope it doesn't do any damages though.
>
> I guess the installation material is inside the tar ball?
>

It is very easy to install. just upload it into Elastix using module
installation interface!!. for further information you can check user manual.

No. you have to download it separately from sourceforge.

>
> Thanks
>
>
> On Tue, Aug 3, 2010 at 2:01 AM, Nasir Iqbal wrote:
>
>> Hi Bruce,
>>
>> We have build an Invoicing module (ICTInovice) for Elastix. It is Free,
>> Open Source, Generate PDF Invoices, and can mail invoices to clients!
>>
>> You can download it from http://sourceforge.net/projects/ictinvoice/
>>
>> Note: Currently ICTInvoice
>> only work with Elastix 1.6
>>
>> Regards
>>
>> On Mon, Aug 2, 2010 at 11:26 PM, bruce bruce  wrote:
>>
>>> Hi Everyone,
>>>
>>> Sorry, if it's not directly related to Asterisk. Some of people on this
>>> list might have PBX deployed for their clients. What software do you use to
>>> invoice them so the invoice looks like a proper telecom invoice maybe?
>>>
>>> Prefer:
>>> -opensource with Windows binary available.
>>> -able to create .pdf invoices rather than printable ones.
>>>
>>> Thanks
>>>
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>>
>>
>>
>> --
>> Nasir Iqbal
>>
>> ICT Innovations
>> http://www.ictinnovations.com/
>>
>>
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>
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Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)

[Philipp von Klitzing]

Also:

There are at least two implementations of the g726 codec, i.e. g726 and 
g726aal2. For this also look at the g726nonstandard setting in sip.conf. 
It is quite possible that your problem is here.

For quick testing to see if the codec works at all: Configure your phones 
to do g726 only (so no alaw/ualaw at all).

Philipp


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Re: [asterisk-users] What do you use for Invoicing?

[Brian C. Huffman]

 I'm still working on it, but I am using a2billing and making 
modifications to some of the PHP code.  I modified the layouts of their 
default invoices and and added PDF creation using dompdf 
(http://code.google.com/p/dompdf/downloads/list).


-b

On 08/03/2010 09:41 AM, Zeeshan Zakaria wrote:


I wanted the same, and so wrote my own. There is none free for this 
purpose.


Zeeshan A Zakaria

--
www.ilovetovoip.com 

On 2010-08-03 9:32 AM, "bruce bruce" > wrote:


Yep, I seen that. That is probably the closet thing but looking at he 
interface it makes me not try to install it. Maybe too complicated. I 
wouldn't want to send customer the whole CDRs but rather a nice Bill 
like the telco sends out.


I am currently toying with NCH Invoicing. Those guys make a software 
for anything and everything.


Thanks,
Bruce

On Tue, Aug 3, 2010 at 9:17 AM, Zeeshan Zakaria > wrote:


>
> I know someone who uses a billing solution called 'freeside',
and is happy with it. Personally I...

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Re: [asterisk-users] What do you use for Invoicing?

[Zeeshan Zakaria]

I wanted the same, and so wrote my own. There is none free for this purpose.

Zeeshan A Zakaria

--
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On 2010-08-03 9:32 AM, "bruce bruce"  wrote:

Yep, I seen that. That is probably the closet thing but looking at he
interface it makes me not try to install it. Maybe too complicated. I
wouldn't want to send customer the whole CDRs but rather a nice Bill like
the telco sends out.

I am currently toying with NCH Invoicing. Those guys make a software for
anything and everything.

Thanks,
Bruce

On Tue, Aug 3, 2010 at 9:17 AM, Zeeshan Zakaria  wrote:

> >
> > I know someone who uses a billing solution called 'freeside', and is
> happy with it. Personally I...
> --
>
>
> > _
> > -- Bandwidth and Colocati...
>


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Re: [asterisk-users] What do you use for Invoicing?

[Steve Totaro]

Most customers I have dealt with, including myself, want CDRs with the bill
via online access, or emailed as an attachment (if not too large).

How can they reconcile their bill without call details?

One vendor I used would send a DVD each month with CDRs because we were
averaging ~15,000 - ~25,000 calls a day, mostly inbound on TFNs, but a good
amount, although relatively nothing for outbound.

Thanks,
Steve Totaro

On Tue, Aug 3, 2010 at 9:27 AM, bruce bruce  wrote:

> Yep, I seen that. That is probably the closet thing but looking at he
> interface it makes me not try to install it. Maybe too complicated. I
> wouldn't want to send customer the whole CDRs but rather a nice Bill like
> the telco sends out.
>
> I am currently toying with NCH Invoicing. Those guys make a software for
> anything and everything.
>
> Thanks,
> Bruce
>
> On Tue, Aug 3, 2010 at 9:17 AM, Zeeshan Zakaria wrote:
>
>> I know someone who uses a billing solution called 'freeside', and is happy
>> with it. Personally I developed my own solution because none could satisfy
>> my needs.
>>
>> Zeeshan A Zakaria
>>
>> --
>> www.ilovetovoip.com
>>
>> On 2010-08-03 2:34 AM,  wrote:
>>
>> Hi,
>>
>>
>> On 08-02-2010 20:55, Gordon Henderson wrote:
>>
>> > I generated invoices with PHP code - it uses a LaTe...
>>
>> well, then i must be a geek too, because i also decided to throw some
>> php code together to generate PDFs from sql.  It was just quicker this
>> way rather than looking and trying a buch of other software.  I'm not
>> sure how other (real) softwares work but since then i'm not spending
>> even a minute with invoices, it's all in crontab.
>>
>> regards
>> adam
>>
>>
>> --
>> _
>> -- Bandwidth and Colocati...
>>
>>
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>>
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>>
>
>
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[asterisk-users] FYI: Seen the 2600Hz announcement?

[Alan Lord (News)]

http://gigaom.com/2010/08/03/2600hz-project/

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Re: [asterisk-users] What do you use for Invoicing?

[bruce bruce]

Yep, I seen that. That is probably the closet thing but looking at he
interface it makes me not try to install it. Maybe too complicated. I
wouldn't want to send customer the whole CDRs but rather a nice Bill like
the telco sends out.

I am currently toying with NCH Invoicing. Those guys make a software for
anything and everything.

Thanks,
Bruce

On Tue, Aug 3, 2010 at 9:17 AM, Zeeshan Zakaria  wrote:

> I know someone who uses a billing solution called 'freeside', and is happy
> with it. Personally I developed my own solution because none could satisfy
> my needs.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-08-03 2:34 AM,  wrote:
>
> Hi,
>
>
> On 08-02-2010 20:55, Gordon Henderson wrote:
>
> > I generated invoices with PHP code - it uses a LaTe...
>
> well, then i must be a geek too, because i also decided to throw some
> php code together to generate PDFs from sql.  It was just quicker this
> way rather than looking and trying a buch of other software.  I'm not
> sure how other (real) softwares work but since then i'm not spending
> even a minute with invoices, it's all in crontab.
>
> regards
> adam
>
>
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Re: [asterisk-users] What do you use for Invoicing?

[bruce bruce]

Oh, you seem to be right on. It's actually an install of Elastix. I will be
testing this for sure. Hope it doesn't do any damages though.

I guess the installation material is inside the tar ball?

Thanks

On Tue, Aug 3, 2010 at 2:01 AM, Nasir Iqbal wrote:

> Hi Bruce,
>
> We have build an Invoicing module (ICTInovice) for Elastix. It is Free,
> Open Source, Generate PDF Invoices, and can mail invoices to clients!
>
> You can download it from http://sourceforge.net/projects/ictinvoice/
>
> Note: Currently ICTInvoice
> only work with Elastix 1.6
>
> Regards
>
> On Mon, Aug 2, 2010 at 11:26 PM, bruce bruce  wrote:
>
>> Hi Everyone,
>>
>> Sorry, if it's not directly related to Asterisk. Some of people on this
>> list might have PBX deployed for their clients. What software do you use to
>> invoice them so the invoice looks like a proper telecom invoice maybe?
>>
>> Prefer:
>> -opensource with Windows binary available.
>> -able to create .pdf invoices rather than printable ones.
>>
>> Thanks
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Nasir Iqbal
>
> ICT Innovations
> http://www.ictinnovations.com/
>
>
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Re: [asterisk-users] What do you use for Invoicing?

[bruce bruce]

I agree but the mentioned software is not opensource.
My conditions clearly included opensource.

On Tue, Aug 3, 2010 at 12:35 AM, Nick Brown  wrote:

>   *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
> *Sent:* Tuesday, 3 August 2010 1:58 PM
> *To:* j...@sunfone.com; Asterisk Users Mailing List - Non-Commercial
> Discussion
>
> *Subject:* Re: [asterisk-users] What do you use for Invoicing?
>
>
>
> Maybe good but the first look brought me to a Pay version. Doesn't satisfy
> the opensource condition.
>
>
>
> thanks,
>
>
>
> Open Source software does not necessarily mean free software.
>
>
>
> Nick.
>
>
>
>
>
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Re: [asterisk-users] What do you use for Invoicing?

[Zeeshan Zakaria]

I know someone who uses a billing solution called 'freeside', and is happy
with it. Personally I developed my own solution because none could satisfy
my needs.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-08-03 2:34 AM,  wrote:

Hi,


On 08-02-2010 20:55, Gordon Henderson wrote:

> I generated invoices with PHP code - it uses a LaTe...
well, then i must be a geek too, because i also decided to throw some
php code together to generate PDFs from sql.  It was just quicker this
way rather than looking and trying a buch of other software.  I'm not
sure how other (real) softwares work but since then i'm not spending
even a minute with invoices, it's all in crontab.

regards
adam


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Re: [asterisk-users] RTP stream not passing through router with port forwarding

[Zeeshan Zakaria]

We all know what are you trying to do, and it is not possible to do, but it
is very impolite and annoying to repost it every few days as a new post with
a slightly different subject. Nobody else does it, and you too please avoid
doing it.

Zeeshan A Zakaria

--
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On 2010-08-03 7:35 AM, "C F"  wrote:

Is asterisk and the SIP device behind the same router?
Most routers will not redirect internal NAT requests. So that if you
are trying to have port forwarding done but the request and the
forwarding destination are on the same interface it won't work.


On 8/3/10, Nasir Javaid  wrote:
> Hi,
>
> I am trying to dial a registe...
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Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)

[Jonas Kellens]

Hello Philipp,

thank you for your answer.


On 08/03/2010 01:21 PM, Philipp von Klitzing wrote:
>> Question 3 :
>> How can I get g726 as first preferred codec ??
>>  
> Which Asterisk version are you using?
>

Using Asterisk 1.4.30

> * check if you have disallow/allow settings in the [general] section of
> sip.conf. Depending on your Asterisk version only the order in [general]
> would be respected, but not the order in the individual sip peer/user
> definition
>

In the [general] section of sip.conf I have :

disallow=all
allow=g726
allow=alaw
allow=g729
allow=gsm

> * look at the variable SIP_CODEC for the inbound (first) call leg, and in
> Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND
>

When I read the value of this variable just before the Dial()-statement, 
it is empty.



Jonas.


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Re: [asterisk-users] chinaroby fxo card - never heard of them

[John Novack]

They seem to have taken over manufacture of cards Digium has discontinued.
I have used several of the TE110 card with success and they are identical

John Novack


asteriskguru asteriskguru wrote:

hi,
I am using this card and IP phone about 6 months. There is no issues 
at all.


Installation procedures are same as Digium  analog card.

Hope it helps,
Ashik

On Tue, Aug 3, 2010 at 6:28 AM, Landy Landy > wrote:


Hello.

I'm looking to buy a FXO card to do some testing with two phone
lines I have at home and was looking in ebay some and found some
cheap ones but, the I've never heard of the brand or manufacturer:
chinaroby. They run for about $99 plus shipping. Have any one used
these? or please recommend one... Money IS an issue.

Thanks.




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Re: [asterisk-users] Caller ID issue

[C F]

In most cases wait(.5) will do. I would not recommend using
answer(2000) as that answers the channel, which means you start
getting billed.

On 8/2/10, Peder  wrote:
>> I am using T1's and didn't think the spill would take that long.
>
>> PRI no, E&M yes.
>
> Some PRI take that long too because the telco sends the name in a followup
> message, not in the initial call setup.
>
>
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Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)

[Philipp von Klitzing]

Hi!

> Question 1 :
> [Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer -
> audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)
> why is combined alaw|g726 and not g726|alaw (reverse) ??

Guess: Here the order presented has no meaning for the order of codec 
negotiation.

> Question 2 :
> why do I see on my Grandstream phone that the codec being used is alaw in
> stead of g726 ??

Because that is what the phone and Asterisk have negotiated. ;-)

> Question 3 :
> How can I get g726 as first preferred codec ??

Which Asterisk version are you using?

* check if you have disallow/allow settings in the [general] section of 
sip.conf. Depending on your Asterisk version only the order in [general] 
would be respected, but not the order in the individual sip peer/user 
definition

* look at the variable SIP_CODEC for the inbound (first) call leg, and in 
Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND

* many Asterisk operators have applied the third party "codec negotiation 
patch"

Philipp


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Re: [asterisk-users] mapping of disconnect reasons

[Philipp von Klitzing]

Hi!

> Is there a way to change the mappings of disconnect reasons to certain
> SIP messages? E.G. I need to change the mapping for SIP 402 "Payment
> Required" from 16 (normal termination) like it is in 1.4.24 to 21
> (call rejected) as defined in RFC 3398.

* if you think the mapping is wrong, then you should open a ticket on the
Asterisk bug tracker

* the mapping can only be changed in the code - which you ahve

* Asterisk 1.8 will allow to read SIP response codes in the dialplan via
{HASH(SIP_CAUSE,)}. Asterisk 1.8 also comes with a
'use_q850_reason' configuration option for generating and parsing, if
available, "Reason: Q.850;cause=".

Philipp


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Re: [asterisk-users] RTP stream not passing through router with port forwarding

[C F]

Is asterisk and the SIP device behind the same router?
Most routers will not redirect internal NAT requests. So that if you
are trying to have port forwarding done but the request and the
forwarding destination are on the same interface it won't work.

On 8/3/10, Nasir Javaid  wrote:
> Hi,
>
> I am trying to dial a registered user via his IP:Port mechanism, but problem
> is that the audio data is not reaching to dialed user. here is the scenario.
>
> caller and callee both are registered at asterisk server. asterisk server is
> on public ip so no port forwarding and natting necessary there. however
> caller and callee both are behind router and there is port forwarding
> enabled and nat=yes, qualify=yes in sip.conf for both users.
>
> callee user name:adf
> callee local ip/port:  192.168.0.10:5678
> callee router ip:   116.79.x.x
>
> when we simply dial callee as Dial(SIP/adf) RTP stream reaches perfectly
> fine to 192.168.0.10 through router and INVITE is sent to local ip through
> router.
>
> INVITE sip:a...@192.168.0.10:5678 SIP/2.0 (asterisk somehow manages to
> contact local ip through router and sends rtp there)
>
> but problem arises when i dial using IP:Port combination like this
>
> Dial(SIP/a...@116.79.x.x:5678)
>
> In this case INVITE is sent to router ip instead of local ip through router.
>
> INVITE sip:a...@116.79.x.x:5678 SIP/2.0   (asterisk sends rtp to router ip
> and not local ip)
>
> Similerly TO header also has same ip as INVITE. I think in IP dial rtp is
> not reaching to local ip through router as INVTE is meant for router ip and
> asterisk does not know where to send rtp stream after sending it to router.
>
> how can this issue be resolved? is there something to be done at router
> confiurations or sip.conf parameters. I have already played with
> nat/qualify/canreinvite/directrtp/externip etc parameters.
>
> regards,
>
> Nasir Javaid
>

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[asterisk-users] sip.conf register in realtime DB

[Jonas Kellens]

Hello list,

scrambling different pieces of info together I've come with the following :

I want to have my "register =>" statements in a MySQL-database, so I've 
made the following table.


table ast_config :
id  1
cat_metric  0
var_metric  0
commented  0
filename  sip.conf
category  general
var_name  register
var_val  username:passw...@sip.provider.net


In ext_config (text file) I have :

sipusers => mysql,AsteriskDB,sip_buddies
sippeers => mysql,AsteriskDB,sip_buddies
sip.conf => mysql,AsteriskDB,ast_config

In sip.conf (text file) I have also :

sip.conf :
rtcachefriends=yes ; Cache realtime friends by adding them 
to the internal list
; just like friends added from the 
config file only on a

; as-needed basis? (yes|no)


After a reload I noticed that the registration came through when I 
executed "sip show registrations". This realtime works.

But I then get a lot of the following messages :
/
[Aug  2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify 
is incompatible with dynamic uncached realtime.  Please either turn 
rtcachefriends on or turn qualify off on peer 'user2'
[Aug  2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify 
is incompatible with dynamic uncached realtime.  Please either turn 
rtcachefriends on or turn qualify off on peer 'user2'/



rtcachefriends is turned on (see above)
qualify is on on every peer (and I want it to stay that way)



Can anyone tell me what I need to configure to get a 100% working example ?!



Kind regards,

Jonas.
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[asterisk-users] Fwd: Stupid Macro question

[unserossi]

Hi all, 



 

I have

 

exten => _X.,1,Macro(dundi-priv,${EXTEN})

exten => _X.,2,DIAL(CAPI/contr1/${EXTEN})

 

Now my problem is, that after hanging up a call, the call is instantly 
re-established using the h-extension which is almost a loop.

I am sure this is a stupid question, but what am I doing wrong?

 

Thanks for advice

Oliver
This might fix you up
exten => _X.,1,Macro(dundi-priv,${EXTEN})
exten => _X.,2,DIAL(CAPI/contr1/${EXTEN})
exten => _x_NOANSWER,1,Dial(Zap/g1/${EXTEN:1}) ;
 
This way the Zap call only occurs on a DUNDI noanswer.


-- 

Thanks, but that is not the problem, DUNDi is answering and is forwarding the 
call to the remote box.

hat works fine.

ut immediately after hanging up the call by the client registered on the remote 
box the call is re-established 

sing the h-extension.

his is my problem. 

ere is the console output. Any tipps are highly appreciated.
Thanks, but that is not the problem, DUNDi is answering and is forwarding the 
call to the remote box.
That works fine.
But immediately after hanging up the call by the client registered on the 
remote box the call is re-established 
using the h-extension.
This is my problem. 
Here is the console output. Any tipps are highly appreciated.

-- Accepting AUTHENTICATED call from 10.28.191.147:
   > requested format = alaw,
   > requested prefs = (alaw|ulaw|gsm),
   > actual format = alaw,
   > host prefs = (alaw|ulaw|gsm),
   > priority = mine
-- Executing [1...@dundi-priv-local:1] Dial("IAX2/iaxuser-829", 
"SIP/1400,30,Ttr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called 1400
-- SIP/1400-006c is ringing
-- SIP/1400-006c answered IAX2/iaxuser-829
-- Executing [...@dundi-priv-local:1] Dial("IAX2/iaxuser-829", 
"SIP/1400,30,Ttr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called 1400
-- SIP/1400-006d is ringing
-- SIP/1400-006d answered IAX2/iaxuser-829
-- Executing [...@dundi-priv-local:1] Dial("IAX2/iaxuser-829", 
"SIP/1400,30,Ttr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called 1400
-- SIP/1400-006e is ringing
-- SIP/1400-006e answered IAX2/iaxuser-829
-- Executing [...@dundi-priv-local:1] Dial("IAX2/iaxuser-829", 
"SIP/1400,30,Ttr") in new stack





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