[asterisk-users] Fax setup T.38 Help needed

[Thorben Jensen]

Hi,

I'm looking for someone who can help us setup Fax with T. 38 on asterisk
10.x.x - We need to be able to do FoIP (Fax over IP) as we have no pstn
lines available.

Do you know how to setup a reliable fax system, then we will pay you to
help us do this.

Regards
Thorben
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Re: [asterisk-users] attended transfer with CEL

[Marek Cervenka]

Dne 20.6.2012 18:40, Marek Cervenka napsal(a):

https://wiki.asterisk.org/wiki/display/AST/CEL+Function
on this wiki is "THIS IS NO LONGER TRUE REWRITE"

is there some way to write userfield,accountcode to the cel?



solved. it's   set(CHANNEL(userfield)=something)

another question
i'm using patch from https://issues.asterisk.org/jira/browse/ASTERISK-18037
it works great

but there is problem(bug?) in second axfer

A - call - B - axfer(AtoC) - C - axfer(AtoD) D

in cel is
eventtype, cid_num, exten
HOLD_START, A, B
HOLD_STOP, A, B
BUT second axfer is
HOLD_START, B, C
HOLD_STOP, B, C

this is strange because on hold is A. is it a bug?

very big problem is that, i cant get info about A - D call (after second 
axfer). there is no info about bridged channel A after axfer



Dne 5.6.2012 13:21, Marek Cervenka napsal(a):

hello,

is there someone who successfully get info about attended transfer 
from CEL?

if yes, can you post some hints/algorithm/...?

scenario
A - customer
B - secretary
C - consultant1
D - consultant2

A -> B
B axfer C
C axfer D

i need to know
time B with C (consultation)
time A with C
time C with D (consultation)
time A with D
time A with everyone (full time -  from start to the end of call)








--
---
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Centrum Vypocetni Techniky
jabber  - cerve...@slu.cz
CVT - http://cvt.fpf.slu.cz
FPF SLU OPAVA   - http://www.fpf.slu.cz
RHCE,RHCVA 100-175-678
===


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Re: [asterisk-users] 10.5.0: channel name inserted as callerid number ??

[Warren Selby]

On Wed, Jun 20, 2012 at 3:21 PM, sean darcy  wrote:

> [home_outgoing]
> type=friend
> transport=tcp
> secret=<>
> fromuser=office_incoming
> host=dynamic
> disallow=all
> allow=ulaw
>


It's because you're using "fromuser" as your username setting.  This will
overwrite your CallerID settings.  Instead try using "defaultuser".


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http://www.SelbyTech.com 
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[asterisk-users] 10.5.0: channel name inserted as callerid number ??

[sean darcy]

I'm trying to set the callerid on a SIP call:

  same=n,Set(CALLERID(all)="test"<2023214321>)
  same=n,Dial(SIP/home_outgoing/150)

-- Executing [202454@from-test-sip:3] Set("SIP/sip-test-0019", 
"CALLERID(all)="test"<2023214321>") in new stack
-- Executing [202454@from-test-sip:4] Dial("SIP/sip-test-0019", 
"SIP/home_outgoing/150") in new stack


[home_outgoing]
type=friend
transport=tcp
secret=<>
fromuser=office_incoming
host=dynamic
disallow=all
allow=ulaw

But the answering box inserts the channel name as the callerid number, 
though the callerid name is correct:


[from_home]
exten => 150,1,NoOp(${CALLERID(all)})

-- Executing [150@from_home:1] NoOp("SIP/office_incoming-0043", 
""test" ") in new stack


[office_incoming]
type=user
transport=tcp
context=from_home
dtmfmode=rfc2833
disallow=all
allow=ulaw

Puzzled. Any help appreciated.

sean


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Re: [asterisk-users] GoogleVoice woes

[Warren Selby]

On Wed, Jun 20, 2012 at 12:30 PM, Chris Gentle  wrote:

> On Wed, Jun 20, 2012 at 12:14 PM, Warren Selby wrote:
>
>> As you said, GV and asterisk integration is unstable at best.  I haven't
>> worked with it in a while, to be honest.  But, with all that being said,
>> I'm not opposed to popping my GV test box back online and helping to
>> troubleshoot.  Why don't you start by giving us the contents of the
>> gtalk.conf and jabber.conf files, the incoming dialplan snippet from
>> extensions.conf for the google voice calls, and the CLI output with
>> verbosity set to at least 6 of both a successful incoming call and a failed
>> incoming call.  There's a debug option of jabber also, if you can have that
>> enabled when you make the calls, that would be very helpful as well.
>>
>
> For the number that is not working there is no jabber debug output and
> nothing shows up on the console.  That leads me to believe that Google
> isn't sending the call to my box at all.
>
>
I remember seeing this when I had my gmail web client open - the call would
try to ring in the web client instead of the asterisk box.  It was
difficult to tell this was the case, because I never really noticed the
ring on the web interface until a few hours into debugging the issue.
However, closing the web app made it ring into the asterisk box.  I'm
assuming you don't have the web client open on a computer somewhere when
you attempt this?  Might be something to check out.


> Here's my gtalk:
>
> [general]
> context=incoming
> allowguest=yes
> bindaddr=0.0.0.0
>
> [guest]
> disallow=all
> allow=ulaw
> context=from-googlevoice
> connection=tcg-asterisk
>
>
Looks pretty similar to my notes on what I had for my own setup, I'll need
to find the config I used on the old box to confirm.



> And my jabber.conf:
>
> [general]
> autoregister=yes
>
> [tcg-asterisk]
> type=client
> serverhost=talk.google.com
> username=my_usern...@gmail.com/Talk
> secret=deleted
> port=5222
> usetls=yes
> usesasl=yes
> statusmessage="Connected via Asterisk"
> timeout=100
>
> [seg-asterisk]
> type=client
> serverhost=talk.google.com
> username=my_wifes_usern...@gmail.com/Talk
> secret=deleted
> port=5222
> usetls=yes
> usesasl=yes
> statusmessage="Connected via Asterisk"
> timeout=100
>
>
This looks pretty close to mine, the only thing I can think to do here
would be to add a "status=available" option to both user definitions, and
also maybe add a buddy= option, and add the name / email of another gmail
user account that you can open in the actual gtalk client.  This will let
you see if these definitions are even coming online at all?

If none of this helps, let me know and I'll find my old GV box and set it
up again.

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http://www.SelbyTech.com 
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[asterisk-users] question on meetme

[Jerry Geis]

I have a meetme running that is taking audio from a PC running asterisk 
(console) as input
to my server that is then feeding it using meetme to two other asterisk 
PC's going out the console.

All running 1.4.43

I have noticed that when the meetme first starts if I change the input 
audio (new song) it changes very fast
at the output to keep up. Over time, like one hour later, if I change 
songs it can be like 5 seconds before

the output changes.

How can I keep that change time low. I'm concerned that over days and 
days it will keep getting worse

and longer.

Thanks,

Jerry

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Re: [asterisk-users] Overview of SIP error codes and possible causes?

[Stefan at WPF]

Thank you Jonathan, I have read up on this, therefore 488 and the
referenced 606 error, but I have to say it wouldn't have helped me. I find
the description still very general. If one looks at the asterisk source
code, then one can clearly find the case with the crypto line and missing
RTP/SAVP. Describing these concrete occurences of errors is what I am
currently missing.

2012/6/20 Jonathan Rose 

> Stefan at WPF wrote:
>
> > is there anywhere an overview of SIP error codes and under which
> > condition they are reported by Asterisk?
> > There are general definitions for SIP error codes, but they are quite
> > general and it's Asterisk that actually checks what's wrong and then
> > reports an error. Now, currently I could check the source code to
> > get more informations what could have caused the error, but that's
> > very time consuming.
> >
> > An example:
> > I recently had the "488 Not acceptable here" error. There were no
> > more details, only this error code. I had no idea what could cause
> > this error (what is not acceptable?) and where to start looking for
> > problems (except maybe check the source code of Asterisk). A
> > documentation of all possible SIP errors and under which conditions
> > they are reported - like the following example - would be very
> > helpful in such cases:
> >
> > Description of "488 Not acceptable here"
> > - Could be caused by codec problems, when codec negotiation failed.
> > You can check if the negotiation failed by 
> > - Can be paused by a phone offering encryption, but only offering
> > RTP/AVP instead of RTP/SAVP profile. Check if the sip log contains a
> > crypto line and only RTP/AVP, if yes, change the phone settings from
> > RTP/AVP to RTP/SAVP or disable RTP encryption in the phone's
> > settings.
> > [Even better: Besides throwing the error message also add the reason
> > for it, at least in the Asterisk log files. I had a warning from
> > Asterisk before the error code, but a warning is still something
> > different than an error, for me the relation between both, the
> > warning and the error message, weren't clear]
> >
> >
> > Is there something like this already? How about introducing it, e.g.
> > every Asterisk developer throwing an error message in his code adds
> > the reason for throwing the error message to an explanation of
> > possible causes, like in the example above?
> >
> > Best regards
> > Stefan
> >
>
> Hi Stefan, it's hardly Asterisk specific, but I'd recommend you
> try RFC 3261 http://www.ietf.org/rfc/rfc3261.txt
>
> In section 21.4, most if not all of the SIP 4XX request errors are
> mentioned including the one you just noted (488).
>
> --
> Jonathan R. Rose
> Digium, Inc. | Software Engineer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> direct +1 256 428 6139
>
> Check us out at: http://digium.com & http://asterisk.org
>
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Re: [asterisk-users] Overview of SIP error codes and possible causes?

[Jonathan Rose]

Stefan at WPF wrote:

> is there anywhere an overview of SIP error codes and under which
> condition they are reported by Asterisk?
> There are general definitions for SIP error codes, but they are quite
> general and it's Asterisk that actually checks what's wrong and then
> reports an error. Now, currently I could check the source code to
> get more informations what could have caused the error, but that's
> very time consuming.
> 
> An example:
> I recently had the "488 Not acceptable here" error. There were no
> more details, only this error code. I had no idea what could cause
> this error (what is not acceptable?) and where to start looking for
> problems (except maybe check the source code of Asterisk). A
> documentation of all possible SIP errors and under which conditions
> they are reported - like the following example - would be very
> helpful in such cases:
> 
> Description of "488 Not acceptable here"
> - Could be caused by codec problems, when codec negotiation failed.
> You can check if the negotiation failed by 
> - Can be paused by a phone offering encryption, but only offering
> RTP/AVP instead of RTP/SAVP profile. Check if the sip log contains a
> crypto line and only RTP/AVP, if yes, change the phone settings from
> RTP/AVP to RTP/SAVP or disable RTP encryption in the phone's
> settings.
> [Even better: Besides throwing the error message also add the reason
> for it, at least in the Asterisk log files. I had a warning from
> Asterisk before the error code, but a warning is still something
> different than an error, for me the relation between both, the
> warning and the error message, weren't clear]
> 
> 
> Is there something like this already? How about introducing it, e.g.
> every Asterisk developer throwing an error message in his code adds
> the reason for throwing the error message to an explanation of
> possible causes, like in the example above?
> 
> Best regards
> Stefan
> 

Hi Stefan, it's hardly Asterisk specific, but I'd recommend you
try RFC 3261 http://www.ietf.org/rfc/rfc3261.txt

In section 21.4, most if not all of the SIP 4XX request errors are
mentioned including the one you just noted (488).

--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct +1 256 428 6139 

Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk

[Stefan at WPF]

Yeah, I noted that too, but besides that it seems like it is exactly what I
am looking for. I am especially confused that there's no hint like "hey,
buy our new product", just EOL. So let's say I am looking for an
alternative to this. And unfortunately I have to add it's for private use
and I therefore need a free solution, which probably restricts the
selection ): Well, anything better than checking logs by hand would be
already a good start :-)

2012/6/20 Tim Nelson 

> - Original Message -
> > - Original Message -
> >
> > > Hello,
> >
> > > 1) I am wondering what is the best practice to monitor if there are
> > > or were problems with SIP calls on my Asterisk box. E.g. how about
> > > a
> > > software that extracts all calls from the /var/log/asterisk/full (I
> > > have permanently enabled verbose 10 and sip debug) log and tells me
> > > on which of them were problems? Checking the logs manually is very
> > > hard, but as SIP is a standardized protocoll, there should be tools
> > > doing that for you? As an example, a person calling me recently got
> > > a 488 Not acceptable error as reply from my Asterisk box. Nothing
> > > came through to my SIP phone, so I didn't know anything about the
> > > call or the problems (which were on his phone btw). I would like to
> > > be informed about such cases, know that there was a call to my
> > > Asterisk box that made problems.
> >
> > > 2) How about monitoring speech quality? E.g. sometimes it seems
> > > like
> > > a packet is missing (I then have a short pause during the call),
> > > how
> > > to monitor such things and create statistics out of this data?
> >
> > > So basically I want to monitor my Asterisk installation proactively
> > > for reliability/problems and (speech) quality.
> >
> >
> > Have a look at VQmonitor:
> >
> > http://www.manageengine.com/products/vqmanager/
> >
> > It works very well.
> >
>
> ...it worked well when you could buy it. Apparently it is EOL now [1].
> Sorry for the noise. These aren't the droids you're looking for.
>
> --Tim
>
> [1] http://www.manageengine.com/products/vqmanager/eol.html
>
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Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk

[Tim Nelson]

- Original Message -
> - Original Message -
> 
> > Hello,
> 
> > 1) I am wondering what is the best practice to monitor if there are
> > or were problems with SIP calls on my Asterisk box. E.g. how about
> > a
> > software that extracts all calls from the /var/log/asterisk/full (I
> > have permanently enabled verbose 10 and sip debug) log and tells me
> > on which of them were problems? Checking the logs manually is very
> > hard, but as SIP is a standardized protocoll, there should be tools
> > doing that for you? As an example, a person calling me recently got
> > a 488 Not acceptable error as reply from my Asterisk box. Nothing
> > came through to my SIP phone, so I didn't know anything about the
> > call or the problems (which were on his phone btw). I would like to
> > be informed about such cases, know that there was a call to my
> > Asterisk box that made problems.
> 
> > 2) How about monitoring speech quality? E.g. sometimes it seems
> > like
> > a packet is missing (I then have a short pause during the call),
> > how
> > to monitor such things and create statistics out of this data?
> 
> > So basically I want to monitor my Asterisk installation proactively
> > for reliability/problems and (speech) quality.
> 
> 
> Have a look at VQmonitor:
> 
> http://www.manageengine.com/products/vqmanager/
> 
> It works very well.
> 

...it worked well when you could buy it. Apparently it is EOL now [1]. Sorry 
for the noise. These aren't the droids you're looking for.

--Tim

[1] http://www.manageengine.com/products/vqmanager/eol.html

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[asterisk-users] Overview of SIP error codes and possible causes?

[Stefan at WPF]

Hello,

is there anywhere an overview of SIP error codes and under which condition
they are reported by Asterisk?
There are general definitions for SIP error codes, but they are quite
general and it's Asterisk that actually checks what's wrong and then
reports an error. Now, currently I could check the source code to get more
informations what could have caused the error, but that's very time
consuming.

An example:
I recently had the "488 Not acceptable here" error. There were no more
details, only this error code. I had no idea what could cause this error
(what is not acceptable?) and where to start looking for problems (except
maybe check the source code of Asterisk). A documentation of all possible
SIP errors and under which conditions they are reported - like the
following example - would be very helpful in such cases:

Description of "488 Not acceptable here"
- Could be caused by codec problems, when codec negotiation failed. You can
check if the negotiation failed by 
- Can be paused by a phone offering encryption, but only offering RTP/AVP
instead of RTP/SAVP profile. Check if the sip log contains a crypto line
and only RTP/AVP, if yes, change the phone settings from RTP/AVP to
RTP/SAVP or disable RTP encryption in the phone's settings.
[Even better: Besides throwing the error message also add the reason for
it, at least in the Asterisk log files. I had a warning from Asterisk
before the error code, but a warning is still something different than an
error, for me the relation between both, the warning and the error message,
weren't clear]


Is there something like this already? How about introducing it, e.g. every
Asterisk developer throwing an error message in his code adds the reason
for throwing the error message to an explanation of possible causes, like
in the example above?

Best regards
Stefan
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Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk

[Tim Nelson]

- Original Message - 

> Hello,

> 1) I am wondering what is the best practice to monitor if there are
> or were problems with SIP calls on my Asterisk box. E.g. how about a
> software that extracts all calls from the /var/log/asterisk/full (I
> have permanently enabled verbose 10 and sip debug) log and tells me
> on which of them were problems? Checking the logs manually is very
> hard, but as SIP is a standardized protocoll, there should be tools
> doing that for you? As an example, a person calling me recently got
> a 488 Not acceptable error as reply from my Asterisk box. Nothing
> came through to my SIP phone, so I didn't know anything about the
> call or the problems (which were on his phone btw). I would like to
> be informed about such cases, know that there was a call to my
> Asterisk box that made problems.

> 2) How about monitoring speech quality? E.g. sometimes it seems like
> a packet is missing (I then have a short pause during the call), how
> to monitor such things and create statistics out of this data?

> So basically I want to monitor my Asterisk installation proactively
> for reliability/problems and (speech) quality.


Have a look at VQmonitor:

http://www.manageengine.com/products/vqmanager/

It works very well.

--Tim

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[asterisk-users] Proactive problem monitoring on SIP on Asterisk

[Stefan at WPF]

Hello,

1) I am wondering what is the best practice to monitor if there are or were
problems with SIP calls on my Asterisk box. E.g. how about a software that
extracts all calls from the /var/log/asterisk/full (I have permanently
enabled verbose 10 and sip debug) log and tells me on which of them were
problems? Checking the logs manually is very hard, but as SIP is a
standardized protocoll, there should be tools doing that for you? As an
example, a person calling me recently got a 488 Not acceptable error as
reply from my Asterisk box. Nothing came through to my SIP phone, so I
didn't know anything about the call or the problems (which were on his
phone btw). I would like to be informed about such cases, know that there
was a call to my Asterisk box that made problems.

2) How about monitoring speech quality? E.g. sometimes it seems like a
packet is missing (I then have a short pause during the call), how to
monitor such things and create statistics out of this data?

So basically I want to monitor my Asterisk installation proactively for
reliability/problems and (speech) quality.

Thanks for any hints!

Best regards
Stefan
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Re: [asterisk-users] GoogleVoice woes

[Chris Gentle]

On Wed, Jun 20, 2012 at 12:14 PM, Warren Selby wrote:

> As you said, GV and asterisk integration is unstable at best.  I haven't
> worked with it in a while, to be honest.  But, with all that being said,
> I'm not opposed to popping my GV test box back online and helping to
> troubleshoot.  Why don't you start by giving us the contents of the
> gtalk.conf and jabber.conf files, the incoming dialplan snippet from
> extensions.conf for the google voice calls, and the CLI output with
> verbosity set to at least 6 of both a successful incoming call and a failed
> incoming call.  There's a debug option of jabber also, if you can have that
> enabled when you make the calls, that would be very helpful as well.
>

For the number that is not working there is no jabber debug output and
nothing shows up on the console.  That leads me to believe that Google
isn't sending the call to my box at all.

Here's my gtalk:

[general]
context=incoming
allowguest=yes
bindaddr=0.0.0.0

[guest]
disallow=all
allow=ulaw
context=from-googlevoice
connection=tcg-asterisk

And my jabber.conf:

[general]
autoregister=yes

[tcg-asterisk]
type=client
serverhost=talk.google.com
username=my_usern...@gmail.com/Talk
secret=deleted
port=5222
usetls=yes
usesasl=yes
statusmessage="Connected via Asterisk"
timeout=100

[seg-asterisk]
type=client
serverhost=talk.google.com
username=my_wifes_usern...@gmail.com/Talk
secret=deleted
port=5222
usetls=yes
usesasl=yes
statusmessage="Connected via Asterisk"
timeout=100

extensions.conf:

;;
; Googlevoice incoming
;;
[from-googlevoice] ;{{{

  ; Google uses call screening even if you have it disabled in your
  ; GoogleVoice profile.  These next two lines sleep for a couple of seconds
  ; and then sends a DTMF 1 digit to accept the call.
  exten => s,1,Wait(2)
  exten => s,n,SendDTMF(1)

  ; Fix Google's crazy caller id string by cutting at the @ and then trim
  ; off the leading +1.
  exten => s,n,Set(crazygooglecid=${CALLERID(name)})
  exten => s,n,Set(stripcrazysuffix=${CUT(crazygooglecid,@,1)})
  exten => s,n,Set(CALLERID(all)=${stripcrazysuffix:2})

  ; Send all incoming calls to [incoming] context
  exten => s,n,Goto(incoming,s,1)
;}}}

-- 
Chris
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Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

[Joseph Towery]

Thanks.  I will go back and use that reference.  I was using examples on web 
pages I was trying to use and just got confused with too much information.





From: Kevin P. Fleming 
To: asterisk-users@lists.digium.com
Sent: Wed, June 20, 2012 10:48:14 AM
Subject: Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

On 06/20/2012 09:34 AM, Joseph Towery wrote:

> Thanks for the tip, the answer is yes, (I forgot I copy the first
> message in into the body below,) but I have read a lot in the
> http://cdn.oreilly.com/books/9780596510480.pdf and
> http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html
> pages. I was just wanting to get the very basic analog config working
> prior to jumping into SIP and other higher level things, and that is
> where I was having a stumbling block. I am making tiny steps forward at
> least right now.
> 

Starting at page 79 of the 2nd Edition of the book, you'll find step-by-step 
instructions on setting up an FXS port for use with an analog telephone.

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

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Re: [asterisk-users] GoogleVoice woes

[Andrew McRory]

I'm with Warren... include your firewall configuration / network setup.

Andrew McRory
Sayso Communications, Inc.
2850 Industrial Plaza
Tallahassee, Florida 32301
Office) 850-224-5737
Mobile) 850-778-3206


On 6/20/2012 1:14 PM, Warren Selby wrote:

On Wed, Jun 20, 2012 at 11:20 AM, Chris Gentle mailto:gent...@gmail.com>> wrote:

I have two GV numbers.  Both are configured to send calls to my
Asterisk 1.8.13.0 box using the Google chat interface.  At one time
I had both working with Asterisk.  Now, for whatever reason, one of
them has stopped sending incoming calls to my asterisk box and
instead just rolls to GV voicemail.  The other number continues to
work fine.  One is associated with my wife's google account and the
other is mine.  I've compared our account settings in Google and
can't find any differences.  Running "jabber show connections" shows
connections to each account.  I know about all the instabilities
with GV and Asterisk but if one number works the other one should
too.  I'm sure this is something simple, probably a Google account
setting that I can't find.  Can anyone think of something else I
might could check?


As you said, GV and asterisk integration is unstable at best.  I haven't
worked with it in a while, to be honest.  But, with all that being said,
I'm not opposed to popping my GV test box back online and helping to
troubleshoot.  Why don't you start by giving us the contents of the
gtalk.conf and jabber.conf files, the incoming dialplan snippet from
extensions.conf for the google voice calls, and the CLI output with
verbosity set to at least 6 of both a successful incoming call and a
failed incoming call.  There's a debug option of jabber also, if you can
have that enabled when you make the calls, that would be very helpful as
well.

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



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Re: [asterisk-users] GoogleVoice woes

[Warren Selby]

On Wed, Jun 20, 2012 at 11:20 AM, Chris Gentle  wrote:

> I have two GV numbers.  Both are configured to send calls to my Asterisk
> 1.8.13.0 box using the Google chat interface.  At one time I had both
> working with Asterisk.  Now, for whatever reason, one of them has stopped
> sending incoming calls to my asterisk box and instead just rolls to GV
> voicemail.  The other number continues to work fine.  One is associated
> with my wife's google account and the other is mine.  I've compared our
> account settings in Google and can't find any differences.  Running "jabber
> show connections" shows connections to each account.  I know about all the
> instabilities with GV and Asterisk but if one number works the other one
> should too.  I'm sure this is something simple, probably a Google account
> setting that I can't find.  Can anyone think of something else I might
> could check?
>

As you said, GV and asterisk integration is unstable at best.  I haven't
worked with it in a while, to be honest.  But, with all that being said,
I'm not opposed to popping my GV test box back online and helping to
troubleshoot.  Why don't you start by giving us the contents of the
gtalk.conf and jabber.conf files, the incoming dialplan snippet from
extensions.conf for the google voice calls, and the CLI output with
verbosity set to at least 6 of both a successful incoming call and a failed
incoming call.  There's a debug option of jabber also, if you can have that
enabled when you make the calls, that would be very helpful as well.

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com 
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Re: [asterisk-users] attended transfer with CEL

[Marek Cervenka]

https://wiki.asterisk.org/wiki/display/AST/CEL+Function
on this wiki is "THIS IS NO LONGER TRUE REWRITE"

is there some way to write userfield,accountcode to the cel?

Dne 5.6.2012 13:21, Marek Cervenka napsal(a):

hello,

is there someone who successfully get info about attended transfer 
from CEL?

if yes, can you post some hints/algorithm/...?

scenario
A - customer
B - secretary
C - consultant1
D - consultant2

A -> B
B axfer C
C axfer D

i need to know
time B with C (consultation)
time A with C
time C with D (consultation)
time A with D
time A with everyone (full time -  from start to the end of call)





--
---
Marek Cervenka
===


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[asterisk-users] GoogleVoice woes

[Chris Gentle]

I have two GV numbers.  Both are configured to send calls to my Asterisk
1.8.13.0 box using the Google chat interface.  At one time I had both
working with Asterisk.  Now, for whatever reason, one of them has stopped
sending incoming calls to my asterisk box and instead just rolls to GV
voicemail.  The other number continues to work fine.  One is associated
with my wife's google account and the other is mine.  I've compared our
account settings in Google and can't find any differences.  Running "jabber
show connections" shows connections to each account.  I know about all the
instabilities with GV and Asterisk but if one number works the other one
should too.  I'm sure this is something simple, probably a Google account
setting that I can't find.  Can anyone think of something else I might
could check?

-- 
Chris
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Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

[Kevin P. Fleming]

On 06/20/2012 09:34 AM, Joseph Towery wrote:


Thanks for the tip, the answer is yes, (I forgot I copy the first
message in into the body below,) but I have read a lot in the
http://cdn.oreilly.com/books/9780596510480.pdf and
http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html
pages. I was just wanting to get the very basic analog config working
prior to jumping into SIP and other higher level things, and that is
where I was having a stumbling block. I am making tiny steps forward at
least right now.



Starting at page 79 of the 2nd Edition of the book, you'll find 
step-by-step instructions on setting up an FXS port for use with an 
analog telephone.


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

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Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

[Joseph Towery]

Yes, I have connected that, and the pci card has the lights on.  I can now lift 
the receiver on the analog phone get dial tone and dial out.  Next I need to 
get 
the phone to ring when called.  Off to do more research.

Thanks for your help.





From: Lyle Giese 
To: asterisk-users@lists.digium.com
Sent: Wed, June 20, 2012 10:12:29 AM
Subject: Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

I have not use a TDM4xx card for a while, but I remember that in order 
for ringing to work, you had to plug in an extra molex connector into 
the card to supply power to the ringing generator portion.

If you forgot to do that...

Lyle

BTW, I know about being a noobie.  I was there once myself and still am 
there every day learning and working with new stuff.  Sometimes not of 
my own choosing, but one must do what they need to keep getting those 
paychecks!

On 6/20/2012 8:44 AM, Joseph Towery wrote:
> Thanks Lyle,
>
> Sorry to sound so much like a newb but in asterisk I am.  I was
> initially trying to do things by hand in the extensions.conf file and
> had no luck.  I then got from SVN checkout asterisk-gui and used it to
> simply try and get things started, and created a trunk, users, incoming
> rule, etc. from the gui and finally got dial tone, and can dial out, but
> I haven't got the analog phone ringing yet.  I will have more targeted
> questions in the near future.  It is just hard to find "google" help for
> analog answers.  Most deal with SIP (which is my next step once I have
> the analog lines working).
>
> Thanks,
>
> 
> *From:* Lyle Giese 
> *To:* asterisk-users@lists.digium.com
> *Sent:* Tue, June 19, 2012 9:29:12 PM
> *Subject:* Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone
>
> An FXO port needs to be connected to dial tone or your PSTN line. And an
> FXS port needs to be connected to the station equipment(ie. a physical
> phone).
>
> The TDM410 is basically a channel bank to Asterisk, so the channel type
> inside Asterisk is FXO to talk to the physical FXS card and FXS to talk
> to the physical FXO port.
>
> Lyle Giese
> LCR Computer Services, Inc.
>
> On 06/18/12 15:08, Joseph Towery wrote:
>> Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24
>> asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12
>> and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do
>> everything by hand) with a TDM410 with 2FXO and 2FXS.  I have my POTS
>> (PTNS) line plugged into port 1 (FXO) and a analog phone connected to
>> port 3 (FXS).  I compiled asterisk with asterisk samples so I realize
>> that may have messed me up.
>>
>> This is all running on Ubuntu Server 12.04.  I have been
>> googling/researching reading the book, etc.  Everything I find is for
>> SIP softphones etc.  I just want to start by getting the asterisk
>> machine to provide dialtone to the analog phone, and ring that phone
>> when I call the PTSN line.
>>
>> I must be missing something in the basic dahdi and dialplan to simple
>> get the analog phone to work.  Can someone point me to a example of
>> what I am trying to accomplish?  Not wanting handholding but a push in
>> the right direction.
>>
>> Thanks.
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided byhttp://www.api-digital.com  --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

[Joseph Towery]

Kevin,
Thanks for the tip, the answer is yes, (I forgot I copy the first message in 
into the body below,) but I have read a lot in the 
http://cdn.oreilly.com/books/9780596510480.pdf and 
http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html pages.  I 
was 
just wanting to get the very basic analog config working prior to jumping into 
SIP and other higher level things, and that is where I was having a stumbling 
block.  I am making tiny steps forward at least right now.  


Thanks





From: Kevin P. Fleming 
To: asterisk-users@lists.digium.com
Sent: Wed, June 20, 2012 10:06:48 AM
Subject: Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

On 06/20/2012 08:44 AM, Joseph Towery wrote:

> Sorry to sound so much like a newb but in asterisk I am. I was initially
> trying to do things by hand in the extensions.conf file and had no luck.
> I then got from SVN checkout asterisk-gui and used it to simply try and
> get things started, and created a trunk, users, incoming rule, etc. from
> the gui and finally got dial tone, and can dial out, but I haven't got
> the analog phone ringing yet. I will have more targeted questions in the
> near future. It is just hard to find "google" help for analog answers.
> Most deal with SIP (which is my next step once I have the analog lines
> working).

Have you read any of the O'Reilly Asterisk books? They will help you learn 
quite 
a lot about Asterisk, and they are available online.

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

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Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

[Lyle Giese]

I have not use a TDM4xx card for a while, but I remember that in order 
for ringing to work, you had to plug in an extra molex connector into 
the card to supply power to the ringing generator portion.


If you forgot to do that...

Lyle

BTW, I know about being a noobie.  I was there once myself and still am 
there every day learning and working with new stuff.  Sometimes not of 
my own choosing, but one must do what they need to keep getting those 
paychecks!


On 6/20/2012 8:44 AM, Joseph Towery wrote:

Thanks Lyle,

Sorry to sound so much like a newb but in asterisk I am.  I was
initially trying to do things by hand in the extensions.conf file and
had no luck.  I then got from SVN checkout asterisk-gui and used it to
simply try and get things started, and created a trunk, users, incoming
rule, etc. from the gui and finally got dial tone, and can dial out, but
I haven't got the analog phone ringing yet.  I will have more targeted
questions in the near future.  It is just hard to find "google" help for
analog answers.  Most deal with SIP (which is my next step once I have
the analog lines working).

Thanks,


*From:* Lyle Giese 
*To:* asterisk-users@lists.digium.com
*Sent:* Tue, June 19, 2012 9:29:12 PM
*Subject:* Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

An FXO port needs to be connected to dial tone or your PSTN line. And an
FXS port needs to be connected to the station equipment(ie. a physical
phone).

The TDM410 is basically a channel bank to Asterisk, so the channel type
inside Asterisk is FXO to talk to the physical FXS card and FXS to talk
to the physical FXO port.

Lyle Giese
LCR Computer Services, Inc.

On 06/18/12 15:08, Joseph Towery wrote:

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24
asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12
and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do
everything by hand) with a TDM410 with 2FXO and 2FXS.  I have my POTS
(PTNS) line plugged into port 1 (FXO) and a analog phone connected to
port 3 (FXS).  I compiled asterisk with asterisk samples so I realize
that may have messed me up.

This is all running on Ubuntu Server 12.04.  I have been
googling/researching reading the book, etc.  Everything I find is for
SIP softphones etc.  I just want to start by getting the asterisk
machine to provide dialtone to the analog phone, and ring that phone
when I call the PTSN line.

I must be missing something in the basic dahdi and dialplan to simple
get the analog phone to work.  Can someone point me to a example of
what I am trying to accomplish?  Not wanting handholding but a push in
the right direction.

Thanks.


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Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

[Kevin P. Fleming]

On 06/20/2012 08:44 AM, Joseph Towery wrote:


Sorry to sound so much like a newb but in asterisk I am. I was initially
trying to do things by hand in the extensions.conf file and had no luck.
I then got from SVN checkout asterisk-gui and used it to simply try and
get things started, and created a trunk, users, incoming rule, etc. from
the gui and finally got dial tone, and can dial out, but I haven't got
the analog phone ringing yet. I will have more targeted questions in the
near future. It is just hard to find "google" help for analog answers.
Most deal with SIP (which is my next step once I have the analog lines
working).


Have you read any of the O'Reilly Asterisk books? They will help you 
learn quite a lot about Asterisk, and they are available online.


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

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Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

[Joseph Towery]

Thanks Lyle,

Sorry to sound so much like a newb but in asterisk I am.  I was initially 
trying 
to do things by hand in the extensions.conf file and had no luck.  I then got 
from SVN checkout asterisk-gui and used it to simply try and get things 
started, 
and created a trunk, users, incoming rule, etc. from the gui and finally got 
dial tone, and can dial out, but I haven't got the analog phone ringing yet.  I 
will have more targeted questions in the near future.  It is just hard to find 
"google" help for analog answers.  Most deal with SIP (which is my next step 
once I have the analog lines working).

Thanks,





From: Lyle Giese 
To: asterisk-users@lists.digium.com
Sent: Tue, June 19, 2012 9:29:12 PM
Subject: Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

An FXO port needs to be connected to dial tone or your PSTN line.  And an 
FXS port needs to be connected to the station equipment(ie. a physical 
phone).

The TDM410 is basically a channel bank to Asterisk, so the channel type 
inside Asterisk is FXO to talk to the physical FXS card and FXS to talk to 
the physical FXO port.

Lyle Giese
LCR Computer Services, Inc.

On 06/18/12 15:08, Joseph Towery wrote: 
Hello, I have a current asterisk 1.8.13.0 asterisk-addons   1.6.24 
asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1   libpri 1.4.12 
and asterisk-gui 2.1.0.rc1 (not trying to use   the gui, want to do 
everything by hand) with a TDM410 with   2FXO and 2FXS.  I have my POTS 
(PTNS) line plugged into port 1   (FXO) and a analog phone connected to 
port 3 (FXS).  I   compiled asterisk with asterisk samples so I realize 
that may   have messed me up.  

>
>This is all running on Ubuntu Server 12.04.  I have been   
>googling/researching reading the book, etc.  Everything I find   is 
>for 
>SIP softphones etc.  I just want to start by getting   the asterisk 
>machine to provide dialtone to the analog phone,   and ring that phone 
>when I call the PTSN line.
>
>I must be missing something in the basic dahdi and dialplan to   
>simple 
>get the analog phone to work.  Can someone point me to   a example of 
>what I am trying to accomplish?  Not wanting   handholding but a push 
>in 
>the right direction.
>
>Thanks.
>
>
>
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Re: [asterisk-users] urgent

[Gorguez Ka]

thank you guys, i just forgot ';' at the end of request!
thx!!!

2012/6/20 SamyGo 

> whats the output when you do this on mysql
>
> *mysql> show databases;*
>
> If there is no database defined then you definitely need to go through the
> installation steps and see if you've missed to create the A2billing
> Database.
>
>
> On Wed, Jun 20, 2012 at 5:59 PM,  wrote:
>
>> did you put the ; at the end of the sql?
>>
>> ** **
>>
>> show tablename;
>>
>> ** **
>>
>> regards.
>>
>> ** **
>>
>> *Von:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *Im Auftrag von *Gorguez Ka
>> *Gesendet:* Mittwoch, 20. Juni 2012 14:52
>> *An:* asterisk-users@lists.digium.com
>> *Betreff:* [asterisk-users] urgent
>>
>> ** **
>>
>> Hello,
>> I would do the billing on Asterisk with A2Billing, but at the
>> configuration of the mysql database when trying to display the table I
>> created with this command: mysql-u root-p mya2billing
>> with
>> mysql> show tablename
>> since there is not even able to show tables
>> I then return
>> -> And nothing else
>> someone could help me solve this?? 
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] urgent

[SamyGo]

whats the output when you do this on mysql

*mysql> show databases;*

If there is no database defined then you definitely need to go through the
installation steps and see if you've missed to create the A2billing
Database.


On Wed, Jun 20, 2012 at 5:59 PM,  wrote:

> did you put the ; at the end of the sql?
>
> ** **
>
> show tablename;
>
> ** **
>
> regards.
>
> ** **
>
> *Von:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *Im Auftrag von *Gorguez Ka
> *Gesendet:* Mittwoch, 20. Juni 2012 14:52
> *An:* asterisk-users@lists.digium.com
> *Betreff:* [asterisk-users] urgent
>
> ** **
>
> Hello,
> I would do the billing on Asterisk with A2Billing, but at the
> configuration of the mysql database when trying to display the table I
> created with this command: mysql-u root-p mya2billing
> with
> mysql> show tablename
> since there is not even able to show tables
> I then return
> -> And nothing else
> someone could help me solve this?? 
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close

[Jakob-Matthias Böttger]

Am 20.06.2012 14:24, schrieb Darren Sessions:

Hi Jakob,

I just finished replying to your direct email (which you can disregard now as 
this seems to be a different problem). I'm pretty sure I know what the issue 
is, but I'll have to get back to you later this evening (my time).

- D


On Jun 20, 2012, at 4:41 AM, Jakob-Matthias Böttger wrote:


Hi,

i am trying to install the just from git cloned app_swift version. Compiling 
works fine. Install as well. But if i try to load the module at Asterisk it 
fails with.

Command 'module load app_swift.so ' failed.
[Jun 20 11:29:51] WARNING[24217]: loader.c:460 load_dynamic_module: Error 
loading module 'app_swift.so': /usr/lib/asterisk/modules/app_swift.so: 
undefined symbol: swift_port_close
[Jun 20 11:29:51] WARNING[24217]: loader.c:850 load_resource: Module 
'app_swift.so' could not be loaded.

My System Informations:

server*CLI> core show version
Asterisk 1.8.13.0 built by root @ server on a x86_64 running Linux on 
2012-06-20 08:55:14 UTC

root@server:~# uname -r
3.2.0-25-generic

root@server:~# ldd /usr/lib/asterisk/modules/app_swift.so
linux-vdso.so.1 =>  (0x7fff6d3ff000)
libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x7f2010a65000)
/lib64/ld-linux-x86-64.so.2 (0x7f2011041000)

root@server:~# cat /etc/ld.so.conf.d/swift.conf
/opt/swift/lib

root@server:~#ldconfig -v | grep swift
/opt/swift/lib:

libswift.so.6 -> libswift.so.6.0
libceplex_de.so.6 -> libceplex_de.so.6.0
libceplang_de.so.6 -> libceplang_de.so.6.0

root@server:~# swift -V

Cepstral Swift v6.0.1, March 2012

Default Voice:  Matthias-8kHzv6.0.0
Language:   German   v5.1.0
Lexicon:unknown  v0.0.0

Concurrency:1 Port(s) Registered
0 Port(s) In Use

Distribution:   No audio distribution license was found.
Saving audio to a file is disabled.

Copyright (C) 2000-20012, Cepstral LLC.


Do You have any Ideas why that won't work?

Best Regards Jakob Böttger


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Thanks for your fast reply.

thats the output of ld /usr/lib/asterisk/modules/app_swift.so

root@server:/opt/swift/lib# ld /usr/lib/asterisk/modules/app_swift.so
ld: warning: cannot find entry symbol _start; not setting start address
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`ast_tvdiff_ms'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`__ast_pthread_mutex_init'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`ast_config_load2'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`swift_port_close'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`swift_port_open'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`ast_unregister_application'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`ast_register_file_version'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`swift_val_string'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`ast_set_write_format'

/usr/lib/asterisk/modules/app_swift.so: undefined reference to `ast_write'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to `ast_tv'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to `ast_samp2tv'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`ast_config_destroy'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`pbx_builtin_setvar_helper'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`swift_engine_open'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`swift_engine_close'

/usr/lib/asterisk/modules/app_swift.so: undefined reference to `ast_log'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to `ast_answer'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`swift_port_set_voice_by_name'

/usr/lib/asterisk/modules/app_swift.so: undefined reference to `ast_waitfor'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`swift_port_set_callback'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`__ast_module_user_remove'
/usr/lib/asterisk/modules/app_swift.so: undefined reference to 
`ast_stopstream'
/usr/lib/asterisk/modules/app_swift.so: un

Re: [asterisk-users] urgent

[B.Tietz]

did you put the ; at the end of the sql?

show tablename;

regards.

Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Gorguez Ka
Gesendet: Mittwoch, 20. Juni 2012 14:52
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] urgent

Hello,
I would do the billing on Asterisk with A2Billing, but at the configuration of 
the mysql database when trying to display the table I created with this 
command: mysql-u root-p mya2billing
with
mysql> show tablename
since there is not even able to show tables
I then return
-> And nothing else
someone could help me solve this??
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[asterisk-users] urgent

[Gorguez Ka]

Hello,
I would do the billing on Asterisk with A2Billing, but at the configuration
of the mysql database when trying to display the table I created with this
command: mysql-u root-p mya2billing
with
mysql> show tablename
since there is not even able to show tables
I then return
-> And nothing else
someone could help me solve this??
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Re: [asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close

[Darren Sessions]

Hi Jakob,

I just finished replying to your direct email (which you can disregard now as 
this seems to be a different problem). I'm pretty sure I know what the issue 
is, but I'll have to get back to you later this evening (my time). 

- D


On Jun 20, 2012, at 4:41 AM, Jakob-Matthias Böttger wrote:

> Hi,
> 
> i am trying to install the just from git cloned app_swift version. Compiling 
> works fine. Install as well. But if i try to load the module at Asterisk it 
> fails with.
> 
> Command 'module load app_swift.so ' failed.
> [Jun 20 11:29:51] WARNING[24217]: loader.c:460 load_dynamic_module: Error 
> loading module 'app_swift.so': /usr/lib/asterisk/modules/app_swift.so: 
> undefined symbol: swift_port_close
> [Jun 20 11:29:51] WARNING[24217]: loader.c:850 load_resource: Module 
> 'app_swift.so' could not be loaded.
> 
> My System Informations:
> 
> server*CLI> core show version
> Asterisk 1.8.13.0 built by root @ server on a x86_64 running Linux on 
> 2012-06-20 08:55:14 UTC
> 
> root@server:~# uname -r
> 3.2.0-25-generic
> 
> root@server:~# ldd /usr/lib/asterisk/modules/app_swift.so
>linux-vdso.so.1 =>  (0x7fff6d3ff000)
>libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x7f2010a65000)
>/lib64/ld-linux-x86-64.so.2 (0x7f2011041000)
> 
> root@server:~# cat /etc/ld.so.conf.d/swift.conf
> /opt/swift/lib
> 
> root@server:~#ldconfig -v | grep swift
> /opt/swift/lib:
> 
>libswift.so.6 -> libswift.so.6.0
>libceplex_de.so.6 -> libceplex_de.so.6.0
>libceplang_de.so.6 -> libceplang_de.so.6.0
> 
> root@server:~# swift -V
> 
> Cepstral Swift v6.0.1, March 2012
> 
> Default Voice:  Matthias-8kHzv6.0.0
> Language:   German   v5.1.0
> Lexicon:unknown  v0.0.0
> 
> Concurrency:1 Port(s) Registered
>0 Port(s) In Use
> 
> Distribution:   No audio distribution license was found.
>Saving audio to a file is disabled.
> 
> Copyright (C) 2000-20012, Cepstral LLC.
> 
> 
> Do You have any Ideas why that won't work?
> 
> Best Regards Jakob Böttger
> 
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Distinctive Ring

[Willian Castello de Alcantara]

Good morning, I'm trying to distinctive ring internal/external the channel bank 
FXS, after some research that has to be checked by dahdi, but I can not use 
 someone could tell me how should I proceed ??

Thanks

[ ]'s

<><><><><><><><><><><><><><><>
Willian Castello de Alcantara
Ensite Telecom
mail: will...@ensite.com.br
msn: williancaste...@yahoo.com.br
skype: williancastello
Fone: (18) 3643-1214
(18) 9143-0691 - (11) 6488-4769
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[asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close

[Jakob-Matthias Böttger]

Hi,

i am trying to install the just from git cloned app_swift version. 
Compiling works fine. Install as well. But if i try to load the module 
at Asterisk it fails with.


Command 'module load app_swift.so ' failed.
[Jun 20 11:29:51] WARNING[24217]: loader.c:460 load_dynamic_module: 
Error loading module 'app_swift.so': 
/usr/lib/asterisk/modules/app_swift.so: undefined symbol: swift_port_close
[Jun 20 11:29:51] WARNING[24217]: loader.c:850 load_resource: Module 
'app_swift.so' could not be loaded.


My System Informations:

server*CLI> core show version
Asterisk 1.8.13.0 built by root @ server on a x86_64 running Linux on 
2012-06-20 08:55:14 UTC


root@server:~# uname -r
3.2.0-25-generic

root@server:~# ldd /usr/lib/asterisk/modules/app_swift.so
linux-vdso.so.1 =>  (0x7fff6d3ff000)
libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x7f2010a65000)
/lib64/ld-linux-x86-64.so.2 (0x7f2011041000)

root@server:~# cat /etc/ld.so.conf.d/swift.conf
/opt/swift/lib

root@server:~#ldconfig -v | grep swift
/opt/swift/lib:

libswift.so.6 -> libswift.so.6.0
libceplex_de.so.6 -> libceplex_de.so.6.0
libceplang_de.so.6 -> libceplang_de.so.6.0

root@server:~# swift -V

Cepstral Swift v6.0.1, March 2012

Default Voice:  Matthias-8kHzv6.0.0
Language:   German   v5.1.0
Lexicon:unknown  v0.0.0

Concurrency:1 Port(s) Registered
0 Port(s) In Use

Distribution:   No audio distribution license was found.
Saving audio to a file is disabled.

Copyright (C) 2000-20012, Cepstral LLC.


Do You have any Ideas why that won't work?

Best Regards Jakob Böttger




smime.p7s
Description: S/MIME Kryptografische Unterschrift
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[asterisk-users] Asterisk 1.8 / sending fax / spandsp

[Thorsten Göllner]

Hi,

I need a fax-send - setup. I read the book "Asterisk The Definitive 
Guide" chapter 19 (fax) and found 2 options listed there.


1) Using spandsp.
2) Using FFA (Digium Fax For Asterisk).

But the book nor any other article I read point out, what the 
differences or drawbacks are.


Does anyone of you have experience with one or both solutions?

We use:
- Asterisk 1.8.13
- Sangoma AFT A104d (germany, E1)
- libpri
- DAHDI 2.6.1

Thanks for any hint.

-Thorsten-

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