[asterisk-users] motif - gv not working today?

[sean darcy]

I had motif working two days ago but now:

Executing [1171@internal:1] Dial("DAHDI/1-1", "Motif/1171") in new stack
[Feb 12 20:56:18] ERROR[7794][C-0001]: chan_motif.c:1762 
jingle_request: Unable to determine endpoint name and target.


motif.conf:

[11XX](!)
transport=google-v1
disallow=all
allow=ulaw

[1171](11XX)
context=incoming-171
connection=gmail1171

xmpp.conf:

[gmail11XX](!)
type=client
serverhost=talk.google.com
priority=1
port=5222
usetls=yes
usesasl=yes
status=available
statusmessage="Not available"
timeout=5

[gmail1171](gmail11XX)
username=gmail1...@gmail.com
secret=gmailsecret

cli:
xmpp show connections
Jabber Users and their status:
[gmail1171] gmail1...@gmail.com - Connected

Have I messed up, or is google voice just not working today?

sean


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Re: [asterisk-users] how to join calls - not barge?

[sean darcy]

On 02/12/2013 05:37 PM, Rusty Newton wrote:

 Original Message -

From: "sean darcy" 




Can I throw A and B into a confbridge and then add C?  Create a new
channel that grabs the A <-> B channel? Or is there a more straight
forward way to do this?



The Asterisk Definitive guide has some good info on what you can do with ConfBridge. That might 
work for you. See "Advanced Conferencing"[1] and "Conferencing with 
ConfBridge()"[2]

Also, there is the Shared Line Appearance stuff in Asterisk[3]. That's a bit 
more confusing, but may help you as well. I'd recommend playing with both to 
really see if they work for your needs.


[1] http://ofps.oreilly.com/titles/9781449332426/asterisk-SysAdmin.html
[2] 
http://ofps.oreilly.com/titles/9781449332426/asterisk-DP-Deeper.html#confbridgeConferencing
[3] http://ofps.oreilly.com/titles/9781449332426/asterisk-DeviceStates.html#SLA



Thanks.

confBridge could work here, but how do I throw an existing bridge into 
the confbridge? That is, if A <-> B exists, how do I trigger the entry 
into confbridge. Once there, it's pretty easy to see how C would join.


maybe EVERY call is done with confbridge. Would that cause some other 
problem?


I don't necessarily have DAHDI, so the SLA stuff wouldn't work. Just as 
well, since my head hurt reading about it.


sean


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Re: [asterisk-users] how to join calls - not barge?

[Rusty Newton]

 Original Message -
> From: "sean darcy" 

> 
> Can I throw A and B into a confbridge and then add C?  Create a new
> channel that grabs the A <-> B channel? Or is there a more straight
> forward way to do this?
> 

The Asterisk Definitive guide has some good info on what you can do with 
ConfBridge. That might work for you. See "Advanced Conferencing"[1] and 
"Conferencing with ConfBridge()"[2]

Also, there is the Shared Line Appearance stuff in Asterisk[3]. That's a bit 
more confusing, but may help you as well. I'd recommend playing with both to 
really see if they work for your needs.


[1] http://ofps.oreilly.com/titles/9781449332426/asterisk-SysAdmin.html  
[2] 
http://ofps.oreilly.com/titles/9781449332426/asterisk-DP-Deeper.html#confbridgeConferencing
[3] http://ofps.oreilly.com/titles/9781449332426/asterisk-DeviceStates.html#SLA


-- 
Rusty Newton 
OS Community Support Manager | Digium, Inc. | www.digium.com  



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Re: [asterisk-users] Problem using ast_tls_cert script

[Daniel - Asterisk]

Hey guys!

I'm sure this question has an answer, surely is not a pure Asterisk matter,
but is important if you can share your thoughts to get TLS working on my
Centos 5.9 server (kernel: 2.6.18-348.1.1.el5).

Again, any hint will be very appreciated!

Elder D. Arohuanca
Lima - Peru


On Thu, Feb 7, 2013 at 2:14 PM, Daniel - Asterisk wrote:

> I did follow instructions in debian without problems, this issue arise
> when trying with Centos 5.8 and 5.9.
>
> On Debian 6.0.6 i wrote:
> ./ast_tls_cert -C 10.200.x.y -O "Company" -d /etc/asterisk/keys/
>
> and I got ca.cert which is working on my Blink phones.
>
> If you have any news please let me know,
>
> Thank you!
>
> Elder
>
>
> On Thu, Feb 7, 2013 at 1:18 PM, kepin sinatra wrote:
>
>> I'm not sure, but it looks like a command in centos and ubuntu are same
>> ...
>> i'am also trying to configure TLS on ubuntu but always error on the
>> softphone blink: transport error.
>>
>>
>> On Fri, Feb 8, 2013 at 12:23 AM, Daniel - Asterisk 
>> wrote:
>>
>>> Hello Kepin,
>>>
>>> I don's know if there's a difference, I changed order with the same
>>> result. Did you find a different script with CentOS?
>>>
>>> Elder
>>>
>>>
>>> On Wed, Feb 6, 2013 at 6:16 PM, kepin sinatra wrote:
>>>
 hi daniel, are you sure the command in debian and ubuntu same?

 On Wed, Feb 6, 2013 at 10:59 PM, Daniel - Asterisk <
 earohua...@gmail.com> wrote:

> Hi List,
>
> I'm trying to set my Asterisk 1.8.20.1 with TLS on CentOS 5.9, it was
> easy and straightforward with Debian 6.0.6, but when I introduce this
> command on CentOS:
>
> #./ast_tls_cert -C 10.200.108.17 -O "MyCompany" -d /etc/asterisk/keys/
>
> I got this error message:
>
> hostname: Unknown host
>
> Same result happens when using server's hostname:
> #./ast_tls_cert -C ast-centos -O "MyCompany" -d /etc/asterisk/keys/
>
> Where 'ast-centos' is the result of 'uname -n'
>
> I've followed instructions from:
>
> http://goalbound.blogspot.com/2012/05/configure-asterisk-18110-on-centos-55.html
> and
> https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
>
> Any hint would be appreciated!
>
> Elder D. Arohuanca
> DCAP
> Lima - Peru
>
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>>>
>>>
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>>
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Re: [asterisk-users] [OT] Mediatrix Euro ISDN hangup problem

[giovanni.v]

On 12/02/2013 17.43, Jean-Denis Girard wrote:

You're right, when someone answers it's not a problem. But if the caller
is sent to voicemail and he hangups, we get 30 seconds disconnect tone.


Yes, I imagined Your problem was born on such scenario.


Yes, I already looked for such a configuration option, but unfortunately
couldn't find any. I hoped someone on the list had experience with a
Mediatrix on Euro ISDN.


Unfortunately I don't have such knowledge, despite the fact some 
collegue suggested to try mediatrix devices I never do seriously because 
by reading their documentation I had the feeling of limited versatility.


If You are not locked into this device I would suggest a Vega gateway 
from Sangoma. Their firmware has a configuration option that does just 
what you need:


---8<--
disc_with_progress=0

O: Disconnect SIP call if disconnect , even if disconnect with progress
1 .. 6000: Enable passage of in-band (audio) information on call 
disconnect – pass media through for a maximum of this number of seconds.

---8<--

... or pass this info to the Mediatrix support so they can be inspired 
by competitors ;-)


Regards, Giovanni

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---
Lightweight++ Business Friendly++ Open++

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Re: [asterisk-users] [OT] Mediatrix Euro ISDN hangup problem

[Jean-Denis Girard]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Le 11/02/2013 12:40, giovanni.v a ￩crit :
> On 11/02/2013 17.01, Jean-Denis Girard wrote:
> I believe the first one will be not a viable option at all, no telco
> will change any important protocol compliance rule on a "per subscriber"
> basis.

Well, in this case the subscriber is also the telco!

> Now forget your gateway for a moment and make a call on an imaginary
> phone connected to your PRI, after that call successfully answered let
> the called party hang up before you do.  What you expect to hear? Sure,
> a disconnect tone... so you will put your phone on hook and the phone
> will send a disconnect immediately.
> 
> Your pri<->gateway<->asterisk should work the same, even if the gateway
> does not send a disconnect immediately the user who started that call
> will hang up at least when hearing the disconnect tone (good feedback
> for humans, no?) and asterisk will send a bye to the sip gateway then
> that one shall initiate a disconnect on the user side.


You're right, when someone answers it's not a problem. But if the caller
is sent to voicemail and he hangups, we get 30 seconds disconnect tone.
Or if the caller is sent to a queue and he hangs up, the agent takes a
call which is already hung up.

> Check also if your gateway allow for customization to remap isdn/q.931
> messages to sip.

Yes, I already looked for such a configuration option, but unfortunately
couldn't find any. I hoped someone on the list had experience with a
Mediatrix on Euro ISDN.

> Sorry, hope you will be able to understand because English isn't my
> native language.

No problem understanding, maybe because English is not my native
language either ;)


Thanks a lot,
- -- 
Jean-Denis Girard

SysNux  Syst￨mes  Linux  en Polyn￩sie fran￧aise
http://www.sysnux.pf/   T￩l: +689 50 10 40 / GSM: +689 79 75 27
-BEGIN PGP SIGNATURE-

iEYEARECAAYFAlEacSkACgkQuu7Rv+oOo/g+cgCeMPAPVplRp2o/QvxnWGdoux5q
BTwAnArY230E3pPU8pJ4/OprCUmax8Gs
=kWdl
-END PGP SIGNATURE-

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[asterisk-users] cisco 7940 and asterisk 11

[Julian Lyndon-Smith]

Ever since we upgraded to asterisk 11 we have had audio problems with
our cisco 7940 phones.

The problems manifest themselves by the conversation turning "robotic"
or into silence (to the extent our agents are saying "hello? hello?"
and the customer is saying "I hear you just fine"

We had to change pedantic=no in sip.conf to allow the phones to register

We are assuming that it is the phone<=>asterisk combination because

a) the call recordings of the conversation are perfect (no noise on
the line, conversation is clear) but it is apparent that the agent
cannot hear the customer sometimes (Hello?)

b) we have replaced the cables and switches between the phones and the pbx

c) we don't have the same problem with Aastra 9133i or Polycom 331 phones

Are there any settings in sip.conf that may help this , or a
particular firmware ? Are there any known audio problems with cisco
7940 and asterisk 11 ?

Many thanks

Julian

-- 
Julian Lyndon-Smith
IT Director, Dot R Limited

"I don’t care if it works on your machine!  We are not shipping your machine!”

The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg

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Re: [asterisk-users] How to install in /usr/local/sbin instead of /usr/sbin ? [SOLVED]

[Olivier]

2013/2/12 Doug Lytle 

> >> non-standard locations such as /usr/local/sbin
>
> If compiling from source, it'd normally be specified by the --prefix
> option:
>
> ./configure --prefix=/usr/local
>
> Doug
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
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>


Using the commands bellow, I could install in /usr/local/sbin
./configure --prefix=/usr/local
make
make install

Thanks for sharing this.
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Re: [asterisk-users] Asterisk Realtime Extension... strange behaviour

[Frank]

Remove the line _X. , and try 3 digits other than 110 112 , let us know 
if it works.


On 2/12/13 5:55 AM, Yves A. wrote:

Hi,

I encountered a strange behaviour using realtime extensions... (on
Asterisk 11.2)

when I use the following static dialplan, everything works as expected..:

[from-sip]
exten =>  110,1,Dial(DAHDI/g0/${EXTEN})
exten =>  112,1,Dial(DAHDI/g0/${EXTEN})
exten => _XXX,1,Dial(SIP/${EXTEN})
exten => _X.,1,Dial(DAHDI/g0/${EXTEN})

will say... if a sip phone calls "110" or "112" the call is routed into
PSTN (german emergency call)
if a sip phone calls any three digit number, the call should be routet
to the corresponding SIP user
and if a sip phone calls any other number the call should be routed into
PSTN... thats ok and works as expected.

when I change to realtime:
[from-sip]
switch => Realtime

and put the diaplan into the database
idcontextextenpriorityappappdata
"1""from-sip""110""1""Dial""DAHDI/g0/${EXTEN}"
"2""from-sip""112""1""Dial""DAHDI/g0/${EXTEN}"
"3""from-sip""_XXX""1""Dial""SIP/${EXTEN}"
"4""from-sip""_X.""1""Dial""DAHDI/g0/${EXTEN}"

only the emergency calls work and any other call goes to DAHDI... I cant
reach any other SIP phone.
Even when swapping the content of the rows 3 and 4 in the database to
idcontextextenpriorityappappdata
"1""from-sip""110""1""Dial""DAHDI/g0/${EXTEN}"
"2""from-sip""112""1""Dial""DAHDI/g0/${EXTEN}"
"3""from-sip""_X.""1""Dial""DAHDI/g0/${EXTEN}"
"4""from-sip""_XXX""1""Dial""SIP/${EXTEN}"

makes no difference...
I thought, using realtime extensions would read the dialplan from top to
bottom, ordered by "id"... but it
seems to be ignored somehow and the extension "_X." catches the calls
before the extensionpattern "_XXX" is reached.

I _could_ avoid this be prefixing "external" numbers with a leading 0
for example... but I dont want to... as I said.. using
static extension via extensions.conf the dialplan works as expected...

Am I missing something?

regards,
yves



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Re: [asterisk-users] How to install in /usr/local/sbin instead of/usr/sbin ?

[Olivier]

2013/2/12 

> If this is the case then doing *“make install DESTDIR=../local/sbin” *should
> install in the /usr/local/sbin directory.
>
No  *“make install DESTDIR=../local/sbin” does work
*

> 
>
> ** **
>
> It looks to be using a relative path starting in /usr/sbin/
>
Yes, it looks like but it doesn't behave as it looks ;-)


> 
>
> ** **
>
> Jacob
>
> ** **
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
> *Sent:* Tuesday, February 12, 2013 6:03 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] How to install in /usr/local/sbin instead
> of/usr/sbin ?
>
> ** **
>
> Hi,
>
> Reading comment in the bottom of
> https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+Asterisk,
> I thought I could install asterisk 11 in non-standard locations such as
> /usr/local/sbin simply typing (from source directory):
> make install DESTDIR=/usr/local/sbin
>
> Doing so seems to install elsewhere
> For instance, make install DESTDIR=/usr/sbin installs runtime asterisk in
> /usr/sbin/usr/sbin directory.
>
> Am I correctly understanding the wiki page ?
> What is the appropriate command ?
>
> Regards
>
>
>
> 
>
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Re: [asterisk-users] How to install in /usr/local/sbin instead of /usr/sbin ?

[Doug Lytle]

>> non-standard locations such as /usr/local/sbin 

If compiling from source, it'd normally be specified by the --prefix option: 

./configure --prefix=/usr/local 

Doug 

-- 
Ben Franklin quote: 

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety." 
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Re: [asterisk-users] How to install in /usr/local/sbin instead of/usr/sbin ?

[Jacob . E . Miles]

If this is the case then doing "make install DESTDIR=../local/sbin"
should install in the /usr/local/sbin directory.

 

It looks to be using a relative path starting in /usr/sbin/

 

Jacob

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, February 12, 2013 6:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to install in /usr/local/sbin instead
of/usr/sbin ?

 

Hi,

Reading comment in the bottom of
https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+Aster
isk, I thought I could install asterisk 11 in non-standard locations
such as /usr/local/sbin simply typing (from source directory):
make install DESTDIR=/usr/local/sbin

Doing so seems to install elsewhere
For instance, make install DESTDIR=/usr/sbin installs runtime asterisk
in /usr/sbin/usr/sbin directory.

Am I correctly understanding the wiki page ?
What is the appropriate command ?

Regards





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[asterisk-users] How to install in /usr/local/sbin instead of /usr/sbin ?

[Olivier]

Hi,

Reading comment in the bottom of
https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+Asterisk,
I thought I could install asterisk 11 in non-standard locations such as
/usr/local/sbin simply typing (from source directory):
make install DESTDIR=/usr/local/sbin

Doing so seems to install elsewhere
For instance, make install DESTDIR=/usr/sbin installs runtime asterisk in
/usr/sbin/usr/sbin directory.

Am I correctly understanding the wiki page ?
What is the appropriate command ?

Regards
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[asterisk-users] Asterisk Realtime Extension... strange behaviour

[Yves A.]

Hi,

I encountered a strange behaviour using realtime extensions... (on 
Asterisk 11.2)


when I use the following static dialplan, everything works as expected..:

[from-sip]
exten =>  110,1,Dial(DAHDI/g0/${EXTEN})
exten =>  112,1,Dial(DAHDI/g0/${EXTEN})
exten => _XXX,1,Dial(SIP/${EXTEN})
exten => _X.,1,Dial(DAHDI/g0/${EXTEN})

will say... if a sip phone calls "110" or "112" the call is routed into 
PSTN (german emergency call)
if a sip phone calls any three digit number, the call should be routet 
to the corresponding SIP user
and if a sip phone calls any other number the call should be routed into 
PSTN... thats ok and works as expected.


when I change to realtime:
[from-sip]
switch => Realtime

and put the diaplan into the database
idcontextextenpriorityappappdata
"1""from-sip""110""1""Dial""DAHDI/g0/${EXTEN}"
"2""from-sip""112""1""Dial""DAHDI/g0/${EXTEN}"
"3""from-sip""_XXX""1""Dial""SIP/${EXTEN}"
"4""from-sip""_X.""1""Dial""DAHDI/g0/${EXTEN}"

only the emergency calls work and any other call goes to DAHDI... I cant 
reach any other SIP phone.

Even when swapping the content of the rows 3 and 4 in the database to
idcontextextenpriorityappappdata
"1""from-sip""110""1""Dial""DAHDI/g0/${EXTEN}"
"2""from-sip""112""1""Dial""DAHDI/g0/${EXTEN}"
"3""from-sip""_X.""1""Dial""DAHDI/g0/${EXTEN}"
"4""from-sip""_XXX""1""Dial""SIP/${EXTEN}"

makes no difference...
I thought, using realtime extensions would read the dialplan from top to 
bottom, ordered by "id"... but it
seems to be ignored somehow and the extension "_X." catches the calls 
before the extensionpattern "_XXX" is reached.


I _could_ avoid this be prefixing "external" numbers with a leading 0 
for example... but I dont want to... as I said.. using

static extension via extensions.conf the dialplan works as expected...

Am I missing something?

regards,
yves



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Re: [asterisk-users] ConfBridge performance problem...?

[Thorsten Göllner]

  
  
Hi again,

I did a try on my asterisk 11.2.1 compiled on Ubuntu 12.04 (64 bit)
with a simple Pentium 4 CPU (Intel(R) Pentium(R) D CPU 2.80GHz). I
connected 5 SIP-Users with a ConfBridge. This is my picture:





Please give a a hint where I can change "you parameters" like
denoise etc. So I will try to change these settings on my box also.
I have no experience with Confbridge.

-Thorsten-

Am 08.02.2013 19:34, schrieb Hristo
  Trendev:


  Hi, the quad-core server is a dedicated asterisk
server. I duplicated the tests on a virtual server (running on
another physical server) only to rule out the possibility of
hardware problem with the first sever.

  

Hristo
  
  

On Fri, Feb 8, 2013 at 11:41 AM,
  Thorsten Göllner  wrote:
  
 Hi,
  
  perhaps it is a problem with your Host-Guest-Setup? Did
  you try the Asterisk-Setup on a dedicated server without
  virtualization?
  
  -Thorsten-
  
  Am 07.02.2013 11:42, schrieb Hristo Trendev:
  
  

  
Hi Thorsten,
  
  
  Thanks for your reply. I did check core show
translations, but the following http://lists.digium.com/pipermail/asterisk-users/2012-November/276132.html
suggests that the values displayed are no longer
representing the computation cost only. However
to answer your question:
  
  
  G722 to SLIN16 cost is 9000, reverse
direction is 6000
  ALAW to SLN16 cost is 17000, reverse
direction is 14500
  
  
  
  G722 to SLN cost is 9600, reverse direction
is 8250
  
  ALAW to SLN cost is 9000, reverse direction
is 6000
  
  
  With regards to the CPU usage per core -
inside the VM, where only one core is available,
the CPU was close to 100% when the problem
started to apear, on the physical server with 4
cores, the cores were evenly loaded at about
30-40%. A single call into the conference
consumed between 10-20% depending on whether I
have denoise enabled or not.
  
  
  There is no dahdi board installed, I only use
the dahdi module for conference timer (note that
the problem is also present with the timerfd

  timing module).
  
  
  BR,
  Hristo


  
  On Wed, Feb 6, 2013 at
1:57 PM, Thorsten Göllner 
wrote:
Did you check
  asterisk -rx "core show translation recalc 10"
  
  Am 06.02.2013 13:56, schrieb Thorsten Göllner:
  

   Sorry - I
just read you alsways checked the cpu
usage. Are all cores at 100%? Is it the
atserisk process which consumes it all?

Am 06.02.2013 13:54, schrieb Thorsten
Göllner:
 Did you watch
  the cpu usage (for example with top)?
  You have a board installed which does
  use dahdi? Did you check the command
  "dahdi_test"?
  Maybe a (performance) problem of the
  software ec?
  
  Am 06.02.2013 11:13, schrieb Hristo
  Trendev:
   Hi,

I have been experimenting with
ConfBridge from the asterisk-11
  

Re: [asterisk-users] target number is busy after some calls

[Muhammad]

yes
here is my CLI output:
Everyone is busy/congested at this time (1:1/0/0)
Executing [s@macro-dialout-trunk:20] NoOp("SIP/105-034b", "Dial failed
for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/105-034b",
"s-BUSY,1") in new stack

I have one truck with 5 SIM defined.
I have to call more times to passed busy tone, actually target mobile
number is not busy.
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Re: [asterisk-users] target number is busy after some calls

[longst]

any trace from CLI, we could take a look? 

Sent from Shitian Long


On Feb 11, 2013, at 4:38 PM, Muhammad  wrote:

> Hi,
> 
> I used Asterisk 1.8 and I have a gsm modem with 8 port.
> When I called target number, gsm modem and asterisk show me one of these 
> ports active. after hangup, the actived port is going to idl status and ready 
> to use. but after some call from extension, when I want to call another 
> number, asterisk gives me Busy status, however all ports are idle and ready 
> to use.
> 
> I think asterisk have to flashed my extension. please let me know what is 
> your idea?
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