Re: [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?

2011-01-09 Thread Steve Edwards

On Fri, 7 Jan 2011, Bruce B wrote:


I want to know each and every parameter's detail that can be included in the 

read=
write=

in manager.conf

Where can I find this? 


0) Try and spell check the subject a bit better. It will make it easier 
for the 'next guy' to search for.


1) Google.

2) The Asterisk source code. Even if you aren't a C programmer grepping 
through the source code can be very productive.


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Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000--
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Re: [asterisk-users] Polarity Reverseal....with analog line

2011-01-09 Thread Edwin Quijada

Can I reverse the polarity from Asterisk to get the call ?
I have 5 days with this and I dont know what to do.

I changed zaptel for dAHDI now I have Dahdi 2.4 and asterisk 1.4.30

TIA

*---* 
*-Edwin Quijada 
*-Developer DataBase 
*-JQ Microsistemas 

*-Soporte PostgreSQL

*-www.jqmicrosistemas.com
*-809-849-8087
*---*





 Date: Wed, 5 Jan 2011 17:04:02 -0500
 From: markm-li...@intellasoft.net
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Polarity Reversealwith analog line
 
 Looks like your telco is sending you polarity reversal on sending you a 
 call.  Which is one of the types of setups for analog lines.l
 
  From your console output it looks like the call was handled just fine 
 other than the 'weird event' notification, which I'm not familiar with.
 
 
 
 On 01/05/2011 11:50 AM, Edwin Quijada wrote:
  Hi !
  I ma having trouble with my PTSN line. When I call to my asterisk I get
  this..
 
  -- Executing [...@from-pstn:3] Hangup(Zap/5-1, ) in new stack
  == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1'
  -- Hungup 'Zap/5-1'
  -- Starting simple switch on 'Zap/5-1'
  [Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17
  (Polarity Reversal)...
  [Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17
  (Polarity Reversal)...
  [Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17
  (Polarity Reversal)...
  -- Executing [...@from-pstn:1] Answer(Zap/5-1, ) in new stack
  -- Executing [...@from-pstn:2] Playback(Zap/5-1, vm-intro) in new stack
  -- Zap/5-1 Playing 'vm-intro' (language 'en')
  [Jan 5 12:45:08] WARNING[2893]: chan_dahdi.c:4550 dahdi_handle_event:
  Ring/Off-hook in strange state 6 on channel 5
  -- Executing [...@from-pstn:3] Hangup(Zap/5-1, ) in new stack
  == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1'
  -- Hungup 'Zap/5-1'
 
  I am using 1.4.30 and zaptel 1.12.
 
  Any cluess?
  *---*
  *-Edwin Quijada
  *-Developer DataBase
  *-JQ Microsistemas
  *-Soporte PostgreSQL
  *-www.jqmicrosistemas.com
  *-809-849-8087
  *---*
 
 
 
 
 
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Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?

2011-01-09 Thread Steve Underwood

On 01/08/2011 03:44 AM, Kevin P. Fleming wrote:

On 01/06/2011 11:34 AM, mgra...@mstvp.com wrote:

We should also be very clear that the Siren codecs are supported on the
Polycom SoundStation conference phones and the VVX-1500 Business Media
Phones. These codecs are not supported in the SoundPoint desk phones.
The SoundPoint series support the more basic G.722 codec in the
IP335/450/550/560/650/670 models.


The SoundPoint IP6000 and IP7000 conference phones (and maybe the 
IP5000, I haven't checked) also support G.722.1 and G.722.1C.


The IP6000 is actually model Polycom recommended for testing when we 
implemented G.722.1.


One of the annoying things about the Polycoms is trying to work out what 
they can do. You have to search quite hard to find which codecs each 
model supports.


Steve


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Re: [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?

2011-01-09 Thread Paul Belanger
On 11-01-07 01:33 PM, Bruce B wrote:
 Where can I find this?
 
manager.conf.sample?

-- 
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twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Benefit of PRI vs SIP trunk calls

2011-01-09 Thread Jim Dickenson
I am running version 1.4.x. Where do I get PRICAUSE? I tried making a call that 
was not answered and I did not see any more information. The dumpchan of 
DADHI/23-1 did not happen as that is in a macro that only gets called for an 
answered call.

I only see this:


Executing [91112223...@empl:8] Dial(SIP/mine-0521, 
Dahdi/G1/111222|60|gM(out-dial)) in new stack
DEBUG[4907]: dsp.c:1682 ast_dsp_set_busy_pattern: dsp busy pattern set to 0,0
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/111222
DEBUG[3188]: chan_dahdi.c:10135 pri_dchannel: Queuing frame from 
PRI_EVENT_PROCEEDING on channel 0/23 span 1
-- DAHDI/23-1 is proceeding passing it to SIP/mine-0521
-- DAHDI/23-1 is ringing
DEBUG[3188]: chan_dahdi.c:1790 dahdi_enable_ec: Echo cancellation already on
-- DAHDI/23-1 answered SIP/mine-0521
-- Executing [...@macro-out-dial:1] DumpChan(DAHDI/23-1, ) in new stack
Dumping Info For Channel: DAHDI/23-1:

Info:
Name=   DAHDI/23-1
Type=   DAHDI
UniqueID=   sys.domain.com-1294514614.2630
CallerID=   9111222
CallerIDName=   (N/A)
DNIDDigits= (N/A)
RDNIS=  (N/A)
State=  Up (6)
Rings=  0
NativeFormat=   0x4 (ulaw)
WriteFormat=0x4 (ulaw)
ReadFormat= 0x4 (ulaw)
1stFileDescriptor=  35
Framesin=   189 
Framesout=  176 
TimetoHangup=   0
ElapsedTime=0h0m4s
Context=macro-out-dial
Extension=  s
Priority=   1
CallGroup=  
PickupGroup=
Application=DumpChan
Data=   (Empty)
Blocking_in=(Not Blocking)

Variables:
MACRO_DEPTH=1
MACRO_PRIORITY=1
MACRO_CONTEXT=from-outside
MACRO_EXTEN=
DIALEDPEERNUMBER=G1/111222
TRANSFERCAPABILITY=SPEECH

DEBUG[4907]: app_macro.c:379 _macro_exec: Executed application: DumpChan
DEBUG[4907]: app_dial.c:1927 dial_exec_full: Macro exited with status 0
DEBUG[4907]: chan_dahdi.c:3464 dahdi_setoption: Set option AUDIO MODE, value: 
ON(1) on DAHDI/23-1
DEBUG[4907]: chan_dahdi.c:3092 dahdi_hangup: Not yet hungup...  Calling hangup 
once with icause, and clearing call
DEBUG[4907]: chan_dahdi.c:3460 dahdi_setoption: Set option AUDIO MODE, value: 
OFF(0) on DAHDI/23-1
-- Hungup 'DAHDI/23-1'
  == Spawn extension (empl, 9111222, 8) exited non-zero on 
'SIP/mine-0521'
-- Executing [...@empl:1] Verbose(SIP/mine-0521, 2|Hangup 
SIP/mine-0521 with cause 16) in new stack
  == Hangup SIP/mine-0521 with cause 16
-- Executing [...@empl:2] DumpChan(SIP/mine-0521, ) in new stack
Dumping Info For Channel: SIP/mine-0521:

Info:
Name=   SIP/mine-0521
Type=   SIP
UniqueID=   sys.domain.com-1294514614.2629
CallerID=   444555
CallerIDName=   Jim Dickenson
DNIDDigits= 9111222
RDNIS=  (N/A)
State=  Up (6)
Rings=  0
NativeFormat=   0x2 (gsm)
WriteFormat=0x2 (gsm)
ReadFormat= 0x2 (gsm)
1stFileDescriptor=  65
Framesin=   248 
Framesout=  253 
TimetoHangup=   0
ElapsedTime=0h0m0s
Context=empl
Extension=  h
Priority=   2
CallGroup=  
PickupGroup=
Application=DumpChan
Data=   (Empty)
Blocking_in=(Not Blocking)

Variables:
DIALSTATUS=ANSWER
DIALEDTIME=5
ANSWEREDTIME=1
RTPAUDIOQOS=ssrc=671389293;themssrc=651772178;lp=0;rxjitter=0.001217;rxcount=248;txjitter=0.00;txcount=252;rlp=0;rtt=0.00
BRIDGEPEER=DAHDI/23-1
DIALEDPEERNUMBER=G1/111222
DIALEDPEERNAME=DAHDI/23-1
MACRO_DEPTH=0
RCStatus=0
MyChan=SIP
sipcallid=0b69233cd5469...@192.168.0.16
SIPUSERAGENT=Grandstream GXP2000 1.2.2.6
SIPDOMAIN=sys.domain.com
SIPURI=sip:m...@00.00.000.000:5064

   -- Executing [...@empl:3] ExecIf(SIP/mine-0521, 
0|Set|DB(conf//haveadmin)=no) in new stack

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

CfMC
http://www.cfmc.com/



On Jan 7, 2011, at 12:44 PM, C F wrote:

 PRICAUSE will give you lots of info on why a call was hungup on. Not
 sure if SIP will give you the same.
 
 On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson dicken...@cfmc.com wrote:
 Does Asterisk, currently using version 1.4, get any more information about 
 the result of an outbound call made over a PRI line compared to a call via a 
 SIP trunk?
 
 As an example, in a PRI call there is this message that shows up on the 
 console:
 
 [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network.
 
 for a call to a fax machine. Does asterisk set anything that a dialplan can 
 access that can know the call was to a fax machine?
 

[asterisk-users] Mail list Woes?

2011-01-09 Thread William Stillwell
Anybody notice log delays in this list, and very small amount of traffic?

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Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?

2011-01-09 Thread mgraves
Actually, all of the conference phones are known by the SoundStation
name and the desk phones are SoundPoint.


Michael Graves
mgraves  mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype mjgraves

  Original Message 
 Subject: Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD
 voice codecs?
 From: Steve Underwood ste...@coppice.org
 Date: Sat, January 08, 2011 11:16 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
 On 01/08/2011 03:44 AM, Kevin P. Fleming wrote:
  On 01/06/2011 11:34 AM, mgra...@mstvp.com wrote:
  We should also be very clear that the Siren codecs are supported on the
  Polycom SoundStation conference phones and the VVX-1500 Business Media
  Phones. These codecs are not supported in the SoundPoint desk phones.
  The SoundPoint series support the more basic G.722 codec in the
  IP335/450/550/560/650/670 models.
 
  The SoundPoint IP6000 and IP7000 conference phones (and maybe the 
  IP5000, I haven't checked) also support G.722.1 and G.722.1C.
 
 The IP6000 is actually model Polycom recommended for testing when we 
 implemented G.722.1.
 
 One of the annoying things about the Polycoms is trying to work out what 
 they can do. You have to search quite hard to find which codecs each 
 model supports.
 
 Steve
 
 
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Re: [asterisk-users] system lockup when going into conference

2011-01-09 Thread Tzafrir Cohen
On Thu, Jan 06, 2011 at 07:01:00PM -0500, cov...@ccs.covici.com wrote:
 Hi.  I have an asterisk system under Debian Leni using asterisk 1.8 with
 no Digium hardware -- and when I go into a meetme conference the system
 either locks up or is 100% cpu utilized or something -- I can't type
 anything and I have to physically reboot the system. The dahdi module is
 loaded and the last log entry is the playing of you are the only person
 in this conference,.
 
 How would I even start to debug this one?
 
 Any ideas would be appreciated.

What version of DAHDI?

-- 
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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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Re: [asterisk-users] Benefit of PRI vs SIP trunk calls

2011-01-09 Thread Doug Lytle

Jim Dickenson wrote:

I am running version 1.4.x. Where do I get PRICAUSE?


NoOP(Hangup Cause: ${HANGUPCAUSE})

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] Definations of READ/WRITE parameters of manager.conf contexts?

2011-01-09 Thread Bruce B
Thanks Paul. That is exactly what I was looking for.

On Sat, Jan 8, 2011 at 2:07 PM, Paul Belanger pabelan...@digium.com wrote:

 On 11-01-07 01:33 PM, Bruce B wrote:
  Where can I find this?
 
 manager.conf.sample?

 --
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?

2011-01-09 Thread Bruce B
You are the sort of person who takes the time to write something totally in
vain. I never understand the sort of mentality people like you have. No need
to respond with an obnoxious comment.
***I was hoping for pointers in asterisk's new wiki or somewhere else that I
may not be aware of.

On Sat, Jan 8, 2011 at 11:27 AM, Steve Edwards asterisk@sedwards.comwrote:

 On Fri, 7 Jan 2011, Bruce B wrote:

  I want to know each and every parameter's detail that can be included in
 the

 read=
 write=

 in manager.conf

 Where can I find this?


 0) Try and spell check the subject a bit better. It will make it easier for
 the 'next guy' to search for.

 1) Google.

 2) The Asterisk source code. Even if you aren't a C programmer grepping
 through the source code can be very productive.

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000
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Re: [asterisk-users] Mail list Woes?

2011-01-09 Thread jonp
Hehe i thought it was me and never commented

Sent from my android device.

-Original Message-
From: William Stillwell will...@stillwellsoft.com
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com
Sent: Sun, 09 Jan 2011 13:47
Subject: [asterisk-users] Mail list Woes?

Anybody notice log delays in this list, and very small amount of traffic?

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Re: [asterisk-users] Mail list Woes?

2011-01-09 Thread Andrew Latham
On Sun, Jan 9, 2011 at 10:27 AM, William Stillwell
will...@stillwellsoft.com wrote:
 Anybody notice log delays in this list, and very small amount of traffic?

There are posts by Kevin F. and others that there is an issue and they
are working on it.

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

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Re: [asterisk-users] system lockup when going into conference

2011-01-09 Thread covici
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:

 On Thu, Jan 06, 2011 at 07:01:00PM -0500, cov...@ccs.covici.com wrote:
  Hi.  I have an asterisk system under Debian Leni using asterisk 1.8 with
  no Digium hardware -- and when I go into a meetme conference the system
  either locks up or is 100% cpu utilized or something -- I can't type
  anything and I have to physically reboot the system. The dahdi module is
  loaded and the last log entry is the playing of you are the only person
  in this conference,.
  
  How would I even start to debug this one?
  
  Any ideas would be appreciated.
 
 What version of DAHDI?

PProblem solved by Shaun fixing a regression in dahdi-trunk.  Thanks to
him its now working.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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[asterisk-users] Call parking question

2011-01-09 Thread Chris Gentle
I've been playing with call parking in Asterisk 1.8.1.  I'm able to park a
call and then pick it back up.  However, on the second attempt, the #72 DTMF
is ignored.  Asterisk just passes that DTMF on to the caller and the call
parking never happens.  Shouldn't I be able to park a call more than once?

-- 
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Re: [asterisk-users] Mail list Woes?

2011-01-09 Thread Tom Rymes
On Jan 9, 2011, at 8:27 AM, William Stillwell wrote:

 Anybody notice log delays in this list, and very small amount of traffic?

I have noticed multiple hour delays between sending messages and seeing them 
back.

Tom

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[asterisk-users] How to check a number online or offline

2011-01-09 Thread Phuong Hoang
Hi all,
Now i want to check a number (channel) online, offline or unreachable on
asterisk but i don`t know to do. Can anyone help me to solve this issue.
Thanks and best regard!
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[asterisk-users] environment variable + res_mysql.conf

2011-01-09 Thread Chandrakant Solanki
Hi All.

I have export some db parameter in /etc/bashrc as follows ...

export DB_NAME=xyz
export DB_IP=1x.1x.1x.1x
export DB_PWD=dkjfaoi

Now, I want use these all environment variable into
/etc/asterisk/res_mysql.conf file.

Is there any way to do this..??

-- 
Regards,

Chandrakant Solanki
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[asterisk-users] OPTIONS Packet is retransmitting continuously

2011-01-09 Thread Nikhil

HI

 In My asterisk OPTIONS packet is retransmitting continuously ,any one 
know the reason for this.I am using asterisk 1.6.1.1.


Thanks
Nikhil

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