date:20070522

Am Montag, den 21.05.2007, 23:16 -0500 schrieb Mike Hammett:
 If it is easy, could you enlighten me?  I have another thread on caller ID
 matching, but I haven't received any positive responses.

In the context where your internal calls usually are handled, like this
(my internal phones have SIP accounts like sip501 with a number between
501 and 599):

exten = _5XX,1,Dial(SIP/sip${EXTEN})

insert lines beforehand that check the caller id against the extension:

exten = _5XX,1,GotoIf($[0${EXTEN}=0${CALLERID(num)}?voicemail,1)
exten = _5XX,2,Dial(SIP/sip${EXTEN})


exten = voicemail,1,VoiceMailMain(${CALLERID(num)},s)

If you want them to have to enter their voicemail password although
calling from their own phone, remove the s (IIRC) - there are docs on
the voip-info.org wiki about all this.

Make sure that noone can fake a callerid when coming into that
context... the nice thing about it having it like this is that users
with a softphone can call voicemail instead of a number.

About the key to be pressed when calling one's own voicebox from abroad:
You can use the voicemail.conf settings exitcontext and operator for
this. I do not currently, so caveat emptor:
**voicemail.conf
exitcontext=voicemailout
operator=yes


**extensions.conf
[voicemailout]
exten = a,1,VoiceMailMain()
exten = o,1,VoiceMailMain()


This way the users should be redirected to the voicemail login prompt
when they press * or 0 during the message (Again: beware, I did not
test this). They will have to enter the voicebox number and pin.

I do not know wether there is a method to get the voicebox number at
this point, such that only the pin needs to be entered. Perhaps setting
a variable (before calling voicemail(123) happens) would do the trick,
but I do not know wether that variable will still exists when jumping to
that voicemailout context.

BR
Anselm

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Re: [asterisk-users] Local SMS how-to.

Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player:
 Hello,
 i just want to activate SMS service between my asterisk local sip
 accounts and between asterisk and local sip accounts. How can i do
 this thin? Also i tried smsq to an account but all i obtained is a
 error message: 
 
 ---Cut Here---
 May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to
 open /var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1:
 Permission denied, deleting
 May 22 13:09:37 WARNING[4829] pbx_spool.c: Failed to scan service
 '/var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1'
 ---And Here---
 
 Is necessary supplementary settings in /etc/asterisk/extensions.conf
 and /etc/asterisk/sip.conf ? Is necessary special module? I checked
 apps_sms.so is already loaded.
 
 Thank you for your support guys.

The SMSq stuff is for landline-type SMS, like those that never became
really popular here in Europe ;-) I do not know of any SIP hardphone
that supports them, but regular analog and ISDN handsets behind a
SIP-to-analog/ISDN gateway work for me.

The point of this SMS transfer method is calling the destination handset
with a certain callerid set (which differs between countries - whatever
number the telco prefers to choose - this can also be configured in the
phone). The phone will not ring but instead immediately answer the call
and receive the short message at 1200bps whatever modem standard they
chose to use.

For sending SMS, the handset will call a similarly telco-provided number
(premium-rate numbers here in Germany - maybe that is the reason for the
lack of popularity of this service) and do that 1200bps talk.

If you still think you can make use of it, make sure to call smsq with
the user id that asterisk is running as. That _might_ already do the
trick. If you do not get it running, ask again - I might have a working
setup somewhere around ;-)

Nevertheless, for me, landline SMS is a PITA. The only great thing is
you can upload Ringtones to Siemens gigaset phones.

BR
Anselm

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[asterisk-users] Dialplan Problem - Outgoing

2007-05-22 Thread Erik Wartusch

Hi,

I have some really disturbing problems with Asterisk 1.4.1 and my dialplan for 
outgoing calls. First of all i switched some weeks ago from * 1.2 (bristuffed 
version ) to this version and in my opinion a lot more troubles arose

For outgoing calls I use a Digium B410P with chan_misdn (before a Junghanns 
QuadBRI with zap).

1) So first thing is, that a user reports to me (highly frequented phone, a 
SNOM 360 with latest FM) that 5 out of 10 calls Asterisk is playing after 
DIAL Nobody is available to take your call at the moment. In which specific 
cases Asterisk jumps to s-NOANSWER? 
I read through the UPGRADE.txt but I didn't find any changes or removals of 
commands, regarding to that.

2) Since the Asterisk change (1.2 -- 1.4 ) -- If the SNOM 360 calls a 
Linksys SPA xxx and also Thomson ST 2030  (internally, SIP) there is no audio 
transmission anymore (just ringing). That was working pretty fine before 
I switched of RTP encryption on the SNOM 360 after a hint from SNOM here on 
the list, but it doesn't work.

Finaly I think such a * version switch must be really carefully done and also 
expecting some troubles.

Kind Regards,

Erik

My first lines for the outgoing context (mainly call state handling):
[outgoing]
exten = s,1,Answer
exten = s,2,Noop()
exten = s,3,Wait,1
exten = s,4,Set(TIMEOUT(digit)=4)
exten = s,5,Set(TIMEOUT(response)=10)
exten = s,6,ChanIsAvail(${OUT}); Is there a free 
channe
l?
exten = s,7,Playtones(pm)

;
; No free channel.
;
exten = s,107,Playback(conf-noempty)
exten = s,108,Hangup

;
; CHANUNAVAIL after Dial(), if there is no
; free line.
;
exten = s-CHANUNAVAIL,1,Playback(all-circuits-busy)
exten = s-CHANUNAVAIL,2,Playback(pls-try-again-later)
exten = s-CHANUNAVAIL,3,Hangup

exten = s-BUSY,1,Playtones(busy)
exten = s-BUSY,2,Hangup

exten = s-CONGESTION,1,Playtones(congestion)
exten = s-CONGESTION,2,Hangup

exten = s-NOANSWER,1,Playback(vm-nobodyavail)
exten = s-NOANSWER,2,Hangup

exten = t,1,Goto(#,1)  ; If they take too 
long,
 give up

exten = i,1,Playback(invalid)  ; That's not valid, 
try
 again
exten = i,2,Hangup


...
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[asterisk-users] SIP Echo

Hello all,

One of our clients reported that they are experiencing echo in SIP calls
(even on internal ones). What do you think could be causing echo in
internal SIP calls?

We're using Polycom telephones, do you think they could be causing it?

Thanks,
Alex

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[asterisk-users] Dial out issues.

2007-05-22 Thread Matt Scott

Dear all.

I have what appears to be a configuration error but I cannot for the life of me 
see what it is. (I am a newbie)
I have searched the wikki and google etc but still none the wiser. Any help 
would be very gratefully received.

Problem:
Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given 
congestion signal as per config, unable to open zap channel. All incoming calls 
work well.

Error Message:
[May 22 11:01:48] WARNING[9179]: channel.c:3024 ast_request: No channel type 
registered for '(Zap'
[May 22 11:01:48] WARNING[9179]: app_dial.c:1090 dial_exec_full: Unable to 
create channel of type '(Zap' (cause 66 - Channel not implemented)

Configs:
[EMAIL PROTECTED] asterisk]# cat sip.conf
[general] 
bindport=5060
bindaddr=0.0.0.0
context=default
disallow=all
allow=gsm
allow=ilbc
allow=ulaw
allow=alaw
srvlookup=yes
;
[400]
type=friend
username=400
host=dynamic
secret=12345
regexten=400
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=400
;
[401]
type=friend
username=401
host=dynamic
secret=12345
regexten=401
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=401
;
[402]
type=friend
username=402
host=dynamic
secret=12345
regexten=402
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=402
;
[410]
type=friend
username=410
host=dynamic
secret=12345
regexten=410
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=410
;
[421]
type=friend
username=421
host=dynamic
secret=12345
regexten=421
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=421
;
[450]
type=friend
username=450
host=dynamic
secret=12345
regexten=450
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=450
;
[451]
type=friend
username=451
host=dynamic
secret=12345
regexten=451
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=451
;
[452]
type=friend
username=452
host=dynamic
secret=12345
regexten=452
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=452
[EMAIL PROTECTED] asterisk]# cat extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
OUTBOUND = Zap/g1
CUSTSERVE1 = SIP/401 ;Teresa
CUSTSERVE2 = SIP/402 ; Louise
;CUSTSERVE3 = SIP/404 ; Helen
QUAD1 = SIP/451 ; Matt
QUAD2 = SIP/452 ; Johan
CUSTSERVE = CUSTSERVE1CUSTSERVE1
;
FSEXT1 = SIP/400 ; Angela
;FSEXT2 = SIP/403 ; Nigel
FSEXT3 = SIP/410 ; Matt
;
;ELLIS = SIP/411
;QUEENS = SIP/412
;FSSHOPS = ELLISQUEENS
;
QUAD = SIP/450
;
LONDONSOLE1 = SIP/421 ; Zoe
;LONDONSOLE2 = SIP/422 ; Laura
;LONDONSOLE = LONDONSOLE1LONDONSOLE2
;
;PRESS1 = SIP/431 ; Lucy
;Press2 = SIP/432 ; Gemma
;PRESSOFFICE = PRESS1PRESS2
;
[macro-oneline]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
;
[macro-oneline1]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
;
[default]
;setupdial out
include = from-pstn
;
;test dialplan
exten = _9xxx,1,SayDigits(${EXTEN:1})
;
;setup the phone extensions
exten = 400,1,Macro(oneline,${FSEXT1})
exten = 401,1,Macro(oneline,${CUSTSERVE1})
exten = 402,1,Macro(oneline,${CUSTSERVE2})
exten = 410,1,Macro(oneline,${FSEXT3})
exten = 421,1,Macro(oneline,${LONDONSOLE1})
exten = 450,1,Macro(oneline,${QUAD})
exten = 451,1,Macro(oneline,${QUAD1})
exten = 452,1,Macro(oneline,${QUAD2})
;
exten = 1000,1,Macro(oneline,${CUSTSERVE})
;exten = 2000,1,Macro(oneline,${FSSHOPS})
;exten = 3000,1,Macro(oneline,${PRESSOFFICE})

;
;setup the dial out via te110p
;exten = _X.,1,SetCIDNum(00)
;
;record new voice files
Exten = 501,1,Wait(2)
Exten = 501,n,Record(/tmp/asterisk-recording:gsm)
Exten = 501,n,Wait(2)
Exten = 501,n,Playback(/tmp/asterisk-recording)
Exten = 501,n,wait(2)
Exten = 501,n,Hangup
;
;goto voicemail
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
;
[from-pstn]
;this is linked to zapata.conf and defines where the ddi points
exten = 00,1,Dial(SIP/401SIP/402,15)
exten = 00,2,Voicemail(1000)
;
exten = 769611,1,Macro(oneline1,${FSEXT1})
exten = 769615,1,Macro(oneline1,${LONDONSOLE1})
;exten = 769616,1,Macro(oneline1,${LONDONSOLE2})
exten = 769636,1,Macro(oneline1,${FSEXT1},${401})
;exten = 769637,1,Macro(oneline1,${NIGEL})
;
exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
exten = _9xxx.,2,Congestion()
exten = _9xxx,102,Congestion()
;
exten = 999,1,Dial,(${OUTBOUND}/999)
exten = ,1,Dial,(${OUTBOUND}/999)
;
[EMAIL PROTECTED] asterisk]# more zapata.conf
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
callwaitingcallerid=yes
restrictcid=no
usecallingpres=no
threewaycalling=yes
callreturn=yes
transfer=yes
cancallforward=yes
echocancelwhenbridged=yes
echocancel=yes
musiconhold=default
rxgain=0.0
txgain=0.0
signalling=pri_cpe
switchtype=euroisdn
immediate=no
overlapdial=yes
pridialplan=unknown
prilocaldialplan=unknown

group=1
context = from-pstn
callerid=asreceived
channel = 1-8

Specs:
New IBM hardware, Intel 4 350mhz 512gig RAM
Digium E1 Card TE110P
Linux Fedcore4
asterisk 1.4
zaptel 1.4
libpri 1.4___
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Re: [asterisk-users] SIP Echo

2007-05-22 Thread William Moore


On 5/22/07, Asterisk [EMAIL PROTECTED] wrote:

Hello all,

One of our clients reported that they are experiencing echo in SIP calls
(even on internal ones). What do you think could be causing echo in
internal SIP calls?

We're using Polycom telephones, do you think they could be causing it?


By nature, pure SIP connections will not generate echo.  This is an
acoustic problem on one or both ends.  If your client is using
headsets, make them get better headsets.  It also might be that they
absolutely love speakerphone and the room's acoustics are bad.
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Re: [asterisk-users] SIP Echo

2007-05-22 Thread Alexandre VERNIOL


Hi,

Could be bandwith or/and latency ... Many causes...


Alex

Asterisk a écrit :

Hello all,

One of our clients reported that they are experiencing echo in SIP calls
(even on internal ones). What do you think could be causing echo in
internal SIP calls?

We're using Polycom telephones, do you think they could be causing it?

Thanks,
Alex

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Re: [asterisk-users] Dial out issues.

2007-05-22 Thread William Moore


On 5/22/07, Matt Scott [EMAIL PROTECTED] wrote:

Dear all.

I have what appears to be a configuration error but I cannot for the life of
me see what it is. (I am a newbie)
I have searched the wikki and google etc but still none the wiser. Any help
would be very gratefully received.

Problem:
Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given
congestion signal as per config, unable to open zap channel. All incoming
calls work well.

Error Message:
[May 22 11:01:48] WARNING[9179]: channel.c:3024 ast_request: No channel type
registered for '(Zap'
[May 22 11:01:48] WARNING[9179]: app_dial.c:1090 dial_exec_full: Unable to
create channel of type '(Zap' (cause 66 - Channel not implemented)


It says channel not implemented.  Did you compile asterisk before or
after compiling and installing libpri?  If you type module load
chan_zap.so, what is the output?
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RE: [asterisk-users] Dial out issues.

2007-05-22 Thread Morgan Gilroy

In your dial lines you have an extrac comma (,)

exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})

should be

exten = _9xxx,1,Dial(${OUTBOUND}/${EXTEN:1})

or

exten = _9xxx,1,Dial,${OUTBOUND}/${EXTEN:1}



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Scott
Sent: 22 May 2007 13:11
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dial out issues.

 

Dear all.

 

I have what appears to be a configuration error but I cannot for the
life of me see what it is. (I am a newbie)

I have searched the wikki and google etc but still none the wiser. Any
help would be very gratefully received.

 

Problem:

Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given
congestion signal as per config, unable to open zap channel. All
incoming calls work well.

 

Error Message:

[May 22 11:01:48] WARNING[9179]: channel.c:3024 ast_request: No channel
type registered for '(Zap'
[May 22 11:01:48] WARNING[9179]: app_dial.c:1090 dial_exec_full: Unable
to create channel of type '(Zap' (cause 66 - Channel not implemented)

 

Configs:

[EMAIL PROTECTED] asterisk]# cat sip.conf
[general] 
bindport=5060
bindaddr=0.0.0.0
context=default
disallow=all
allow=gsm
allow=ilbc
allow=ulaw
allow=alaw
srvlookup=yes
;
[400]
type=friend
username=400
host=dynamic
secret=12345
regexten=400
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=400
;
[401]
type=friend
username=401
host=dynamic
secret=12345
regexten=401
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=401
;
[402]
type=friend
username=402
host=dynamic
secret=12345
regexten=402
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=402
;
[410]
type=friend
username=410
host=dynamic
secret=12345
regexten=410
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=410
;
[421]
type=friend
username=421
host=dynamic
secret=12345
regexten=421
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=421
;
[450]
type=friend
username=450
host=dynamic
secret=12345
regexten=450
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=450
;
[451]
type=friend
username=451
host=dynamic
secret=12345
regexten=451
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=451
;
[452]
type=friend
username=452
host=dynamic
secret=12345
regexten=452
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=452
[EMAIL PROTECTED] asterisk]# cat extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
OUTBOUND = Zap/g1
CUSTSERVE1 = SIP/401 ;Teresa
CUSTSERVE2 = SIP/402 ; Louise
;CUSTSERVE3 = SIP/404 ; Helen
QUAD1 = SIP/451 ; Matt
QUAD2 = SIP/452 ; Johan
CUSTSERVE = CUSTSERVE1CUSTSERVE1
;
FSEXT1 = SIP/400 ; Angela
;FSEXT2 = SIP/403 ; Nigel
FSEXT3 = SIP/410 ; Matt
;
;ELLIS = SIP/411
;QUEENS = SIP/412
;FSSHOPS = ELLISQUEENS
;
QUAD = SIP/450
;
LONDONSOLE1 = SIP/421 ; Zoe
;LONDONSOLE2 = SIP/422 ; Laura
;LONDONSOLE = LONDONSOLE1LONDONSOLE2
;
;PRESS1 = SIP/431 ; Lucy
;Press2 = SIP/432 ; Gemma
;PRESSOFFICE = PRESS1PRESS2
;
[macro-oneline]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
;
[macro-oneline1]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
;
[default]
;setupdial out
include = from-pstn
;
;test dialplan
exten = _9xxx,1,SayDigits(${EXTEN:1})
;
;setup the phone extensions
exten = 400,1,Macro(oneline,${FSEXT1})
exten = 401,1,Macro(oneline,${CUSTSERVE1})
exten = 402,1,Macro(oneline,${CUSTSERVE2})
exten = 410,1,Macro(oneline,${FSEXT3})
exten = 421,1,Macro(oneline,${LONDONSOLE1})
exten = 450,1,Macro(oneline,${QUAD})
exten = 451,1,Macro(oneline,${QUAD1})
exten = 452,1,Macro(oneline,${QUAD2})
;
exten = 1000,1,Macro(oneline,${CUSTSERVE})
;exten = 2000,1,Macro(oneline,${FSSHOPS})
;exten = 3000,1,Macro(oneline,${PRESSOFFICE})

 

;
;setup the dial out via te110p
;exten = _X.,1,SetCIDNum(00)
;
;record new voice files
Exten = 501,1,Wait(2)
Exten = 501,n,Record(/tmp/asterisk-recording:gsm)
Exten = 501,n,Wait(2)
Exten = 501,n,Playback(/tmp/asterisk-recording)
Exten = 501,n,wait(2)
Exten = 501,n,Hangup
;
;goto voicemail
exten=*98,1,VoiceMailMain([EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] })
;
[from-pstn]
;this is linked to zapata.conf and defines where the ddi points
exten = 00,1,Dial(SIP/401SIP/402,15)
exten = 00,2,Voicemail(1000)
;
exten = 769611,1,Macro(oneline1,${FSEXT1})
exten = 769615,1,Macro(oneline1,${LONDONSOLE1})
;exten = 769616,1,Macro(oneline1,${LONDONSOLE2})
exten = 769636,1,Macro(oneline1,${FSEXT1},${401})
;exten = 769637,1,Macro(oneline1,${NIGEL})
;
exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
exten = _9xxx.,2,Congestion()
exten = _9xxx,102,Congestion()
;
exten = 999,1,Dial,(${OUTBOUND}/999)
exten = ,1,Dial,(${OUTBOUND}/999)
;
[EMAIL PROTECTED] asterisk]# more zapata.conf
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
callwaitingcallerid=yes
restrictcid=no
usecallingpres=no
threewaycalling=yes

Re: [asterisk-users] Dial out issues.

2007-05-22 Thread William Moore


On 5/22/07, Morgan Gilroy [EMAIL PROTECTED] wrote:

In your dial lines you have an extrac comma (,)

exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})

should be

exten = _9xxx,1,Dial(${OUTBOUND}/${EXTEN:1})

or

exten = _9xxx,1,Dial,${OUTBOUND}/${EXTEN:1}


Good catch Morgan!
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RE: [asterisk-users] SIP Echo

Thank you.

How do you think would be the best way to approach this problem? Do you
think anything else could also produce echo as well?



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
Moore
Sent: Tuesday, May 22, 2007 2:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo

On 5/22/07, Asterisk [EMAIL PROTECTED] wrote:
 Hello all,

 One of our clients reported that they are experiencing echo in SIP
calls
 (even on internal ones). What do you think could be causing echo in
 internal SIP calls?

 We're using Polycom telephones, do you think they could be causing it?

By nature, pure SIP connections will not generate echo.  This is an
acoustic problem on one or both ends.  If your client is using
headsets, make them get better headsets.  It also might be that they
absolutely love speakerphone and the room's acoustics are bad.
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RE: [asterisk-users] SIP Echo

How could I check if bandwith or/and latency is causing it?

If I do SIP show peers it says OK (13 ms) for all peers. I guess there is a way 
to gather more detailed info on SIP calls and latency?

* box is connected to the 1Gb switch with 1Gb connection, and clients have 
100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP hardphones 
connected to the * box.

Thanks, Alex

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexandre VERNIOL
Sent: Tuesday, May 22, 2007 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo

Hi,

Could be bandwith or/and latency ... Many causes...


Alex

Asterisk a écrit :
 Hello all,

 One of our clients reported that they are experiencing echo in SIP calls
 (even on internal ones). What do you think could be causing echo in
 internal SIP calls?

 We're using Polycom telephones, do you think they could be causing it?

 Thanks,
 Alex

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Re: [asterisk-users] SIP Echo

2007-05-22 Thread David Gomillion


We experience echo too from time to time. It's usually headset-related, but
not always. I ran a persistent ping on one of the phones, and we diagnosed a
wiring problem with it. Other phones needed a new handset. The problem is
that these problems need to be fixed on the phone NOT hearing echo.

On 5/22/07, Asterisk [EMAIL PROTECTED] wrote:


How could I check if bandwith or/and latency is causing it?

If I do SIP show peers it says OK (13 ms) for all peers. I guess there is
a way to gather more detailed info on SIP calls and latency?

* box is connected to the 1Gb switch with 1Gb connection, and clients have
100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP hardphones
connected to the * box.

Thanks, Alex

-Original Message-
From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] On Behalf Of Alexandre VERNIOL
Sent: Tuesday, May 22, 2007 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo

Hi,

Could be bandwith or/and latency ... Many causes...


Alex

Asterisk a écrit :
 Hello all,

 One of our clients reported that they are experiencing echo in SIP calls
 (even on internal ones). What do you think could be causing echo in
 internal SIP calls?

 We're using Polycom telephones, do you think they could be causing it?

 Thanks,
 Alex

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[asterisk-users] how can I catch the event generated when a parked call is hung up?

2007-05-22 Thread lavarini

Hi all,

how can I catch the event generated when a parked call is hung up?

In my dialplan, when arrives a call to a specific number, Asterisk parks
the call and announces the parking slot to a number. But if the user hangs
up the parked call, I don't know how to catch the event, from dialplan,
that removes the call from the parking slot. I want to know if there is a
method for do this.

Thank you!

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RE: [asterisk-users] Caller ID matching

2007-05-22 Thread Mike Hammett

Yeah, I was trying to have it match the caller ID with what they're dialing
so that I don't have a separate entry for every customer.

 

 

-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com

 

 

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham
Sent: Tuesday, May 22, 2007 5:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID matching

 

I did it anyway. i used another way around to do it:

suppose 88777 is your number

exten= 88777,1,Dial(SIP/you)
exten= 88777/88777,1,VoiceMailMain()

but in this case you will have to make a separate vm extension for every
user. 

On 5/22/07, Rizwan Hisham [EMAIL PROTECTED] wrote:

well i have tried to solve your problem, making your extensions in my
dialplan and reloading dialplan gives me segmentation fault. im afraid i
cant help u :)

 

exten = 555*,1,NoOp(${CALLERID(num)})

exten = 555*,2,Hangup

 

On 5/20/07, Mike Hammett  mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:

What's going on here?  555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.

 

I'm trying to emulate cell phone voicemail where you call your own number to
check your voicemail.

 

-- Accepting AUTHENTICATED call from 65.182.165.XXX:

requested format = gsm,

requested prefs = (),

actual format = ulaw,

host prefs = (ulaw),

priority = mine

-- Executing NoOp(IAX2/815748-16, 815748) in new stack

-- Executing Hangup(IAX2/815748-16, ) in new stack

  == Spawn extension (outbound-scripted, 555*, 2) exited non-zero on
'IAX2/815748-16'

May 20 11:10:34 ERROR[3286]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql:
cannot connect to database server localhost.

-- Hungup 'IAX2/815748-16'

May 20 11:11:00 NOTICE[3275]: chan_iax2.c:7323 socket_read: Rejected connect
attempt from 65.182.165.XXX, request '[EMAIL PROTECTED]' does not
exist

 

exten = ${CALLERID(NUM)}/${CALLERID(NUM)},1,Answer

exten = ${CALLERID(NUM)}/${CALLERID(NUM)},2,NoOp(It's here)

exten = ${CALLERID(NUM)}/${CALLERID(NUM)},3,VoicemailMain(${CALLERID(NUM)})

exten = ${CALLERID(NUM)}/${CALLERID(NUM)},4,Hangup()

 

exten = 555*,1,NoOp(${CALLERID(num)})

exten = 555*,2,Hangup

 

 

 

-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com

 

 

 

 

 


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-- 
Rizwan Hisham
Software Engineer
AXVOICE Inc. 




-- 
Rizwan Hisham
Software Engineer
AXVOICE Inc. 

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RE: [asterisk-users] SIP Echo

Thanks. I will try to ping my phones, to see what's the situation.

  _  

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gomillion
Sent: Tuesday, May 22, 2007 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo

We experience echo too from time to time. It's usually headset-related, but not 
always. I ran a persistent ping on one of the phones, and we diagnosed a wiring 
problem with it. Other phones needed a new handset. The problem is that these 
problems need to be fixed on the phone NOT hearing echo. 

On 5/22/07, Asterisk [EMAIL PROTECTED] wrote:

How could I check if bandwith or/and latency is causing it?

If I do SIP show peers it says OK (13 ms) for all peers. I guess there is a way 
to gather more detailed info on SIP calls and latency?

* box is connected to the 1Gb switch with 1Gb connection, and clients have 
100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP hardphones 
connected to the * box. 

Thanks, Alex

-Original Message-
From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] 
On Behalf Of Alexandre VERNIOL
Sent: Tuesday, May 22, 2007 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo 

Hi,

Could be bandwith or/and latency ... Many causes...

Alex

Asterisk a écrit :
 Hello all,

 One of our clients reported that they are experiencing echo in SIP calls
 (even on internal ones). What do you think could be causing echo in
 internal SIP calls?

 We're using Polycom telephones, do you think they could be causing it?

 Thanks,
 Alex 

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Re: [asterisk-users] Re: DUNDi configuration problem

2007-05-22 Thread Tim Verscheure


ok so now I changed ext-local to dundi-ext and I created this
context at the bottom of the extensions file. This is now the case.

[dundi-priv-canonical]
; Direct numbers

exten = 5010,1,NooP(DUNDI LOOKUP 5010)
exten = 5011,1,NooP(DUNDI LOOKUP 5011)

exten = _60XX,1,Goto(dundi-ext,${EXTEN},1)

[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
exten = _60XX,1,Goto(dundi-ext,${EXTEN},1)

[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here

[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn

exten = 5010,1,Dial(SIP/5010)
exten = 5011,1,Dial(SIP/5011)

[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv

[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch

[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-lookup

[trydundi]
exten = _.,1,Macro(dundi-priv,${EXTEN})
exten = _.,2,Congestion


This is the dundi-ext at the bottom. In there I put this line:
[dundi-ext]
exten = _60XX,1,Dial(SIP/[EMAIL PROTECTED])

I made this myself, I think that if I get an incoming call from for
example 6010, the person would be dialing SIP/[EMAIL PROTECTED], right?

this is the output:
*CLI -- Executing [EMAIL PROTECTED]:1] Goto(SIP/6010-0820cdc8,
dundi-ext|5011|1) in new stack
   -- Goto (dundi-ext,5011,1)
   -- Executing [EMAIL PROTECTED]:1] Dial(SIP/6010-0820cdc8,
SIP/[EMAIL PROTECTED]) in new stack
[May 22 14:36:18] WARNING[4520]: chan_sip.c:2738 create_addr: No such host: priv
[May 22 14:36:18] WARNING[4520]: app_dial.c:1099 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
 == Everyone is busy/congested at this time (1:0/0/1)
 == Auto fallthrough, channel 'SIP/6010-0820cdc8' status is 'CHANUNAVAIL'


2007/5/21, Remco Post [EMAIL PROTECTED]:

Tim Verscheure wrote:
 Now I get this... If I call from 5011 on the 192.168.1.103 machine to
 6010 on the 192.168.1.69 machine my X-lite softphone says, call
 declined

 this is the output:
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/5011-081da508,
 ext-local|6010|1) in new stack
-- Goto (ext-local,6010,1)
 [May 21 15:32:46] WARNING[8939]: pbx.c:2450 __ast_pbx_run: Channel
 'SIP/5011-081da508' sent into invalid extension '6010' in context
 'ext-local', but no invalid handler


so, is there an extension 6010 in you context ext-local? Probably not ;-)

 I'll add my extension file so you can see it. greetz

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[asterisk-users] Kernel Panic in wct4xxp during unload on Zaptel-1.4.4

2007-05-22 Thread James FitzGibbon


I attempted an upgrade of our production system from Asterisk/Zaptel 1.2 to
1.4 this weekend.  Intially everything looked like it was working properly,
but some time in the day following the upgrade, the system died to a kernel
panic.  I wasn't able to catch the entire kernel dump on the console
unfortunately.

I attempted to isolate the panic, and found that when 'service zaptel stop'
was run (specifically, when wct4xxp was unloaded) I get this panic
consistently:

5 Not prepped yet! (repeated approx 250 times)
5 Freed a Wildcard
5 Not prepped yet! (repeated approx 550 times)
5 Stopped TE4XXP, Turned off DMA
5 Not prepped yet! (repeated approx 11000 times)
5 Unable to handle kernel paging request at ff034010 RIP:
5 a0163207{:wct4xxp:t4_interrupt_gen2+63}
5 PML4 4ea063 PGD 1388067 PMD 1389067 PTE 0
5 Oops:  [1] SMP
5 CPU 1
5 Modules linked in: zttranscode(U) wct4xxp(U) zaptel(U) crc_ccitt
netconsole md5 ipv6 dm_mirror dm_mod button battery ac joydev ehci_hcd
uhci_hcd hw_random bnx2 ext3 jbd
cciss sd_mod scsi_mod
5 Pid: 11053, comm: hotplug Not tainted 2.6.9-42.0.8.ELsmp
5 RIP: 0010:[a0163207]
a0163207{:wct4xxp:t4_interrupt_gen2+63}
5 RSP: :0100013ebdb0  EFLAGS: 00010046
5 RAX: ff034000 RBX: 010073478724 RCX: 0002
5 RDX: 010073478680 RSI: 0002 RDI: 010073478724
5 RBP: 010073478680 R08: 0008 R09: 
5 R10:  R11: 0002 R12: 00d1
5 R13: 0100013ebec8 R14: 0100013ebec8 R15: 010075e94978
5 FS:  002a955643e0() GS:804e5900()
knlGS:
5 CS:  0010 DS:  ES:  CR0: 8005003b
5 CR2: ff034010 CR3: 013d8000 CR4: 06e0
5 Process hotplug (pid: 11053, threadinfo 01007413c000, task
01007c445030)
5 Stack: 00d1 01007413dc98 80138552
0038
50100013ebea8 0100013ebde8 0001
00d1
50012 010073478680
5 Call Trace:IRQ 80138552{printk+141}
80112f4a{handle_IRQ_event+41}
5801131c4{do_IRQ+197}
80110833{ret_from_intr+0}
58013c731{__do_softirq+77}
8013c7e5{do_softirq+49}
580110bf5{apic_timer_interrupt+133}  EOI
8011c21a{flush_tlb_page+44}
580169106{do_wp_page+1127}
80123ed3{do_page_fault+575}
580169ff2{handle_mm_fault+1228}
80123e9a{do_page_fault+518}
58011026a{system_call+126}
80132bc6{schedule_tail+202}
580110d91{error_exit+0}
5
5 Code: 8b 40 10 89 44 24 58 e8 3d 80 1a e0 31 c0 f6 44 24 58 07 0f
5 RIP a0163207{:wct4xxp:t4_interrupt_gen2+63} RSP
0100013ebdb0
5 CR2: ff034010
5  0Kernel panic - not syncing: Oops
5  Badness in panic at kernel/panic.c:118
5
5 Call Trace:IRQ 80137a8a{panic+527}
80110bf5{apic_timer_interrupt+133}
580111aec{oops_end+38} 80111b07{oops_end+65}
580124148{do_page_fault+1204}
a0078f51{:bnx2:bnx2_start_xmit+470}
5802bb4cd{netpoll_send_skb+257}
80110d91{error_exit+0}
5a0163207{:wct4xxp:t4_interrupt_gen2+63}
80138552{printk+141}
580112f4a{handle_IRQ_event+41}
801131c4{do_IRQ+197}
580110833{ret_from_intr+0}
8013c731{__do_softirq+77}
58013c7e5{do_softirq+49}
80110bf5{apic_timer_interrupt+133}
5 EOI 8011c21a{flush_tlb_page+44}
80169106{do_wp_page+1127}
580123ed3{do_page_fault+575}
80169ff2{handle_mm_fault+1228}
580123e9a{do_page_fault+518}
8011026a{system_call+126}
580132bc6{schedule_tail+202}
80110d91{error_exit+0}
5
5 Badness in i8042_panic_blink at drivers/input/serio/i8042.c:987
5
5 Call Trace:IRQ 8024219b{i8042_panic_blink+238}
80137a38{panic+445}
580110bf5{apic_timer_interrupt+133}
80111aec{oops_end+38}
580111b07{oops_end+65}
80124148{do_page_fault+1204}
5a0078f51{:bnx2:bnx2_start_xmit+470}
802bb4cd{netpoll_send_skb+257}
580110d91{error_exit+0}
a0163207{:wct4xxp:t4_interrupt_gen2+63}
580138552{printk+141}
80112f4a{handle_IRQ_event+41}
5801131c4{do_IRQ+197}
80110833{ret_from_intr+0}
58013c731{__do_softirq+77}
8013c7e5{do_softirq+49}
580110bf5{apic_timer_interrupt+133}  EOI
8011c21a{flush_tlb_page+44}
580169106{do_wp_page+1127}
80123ed3{do_page_fault+575}
580169ff2{handle_mm_fault+1228}
80123e9a{do_page_fault+518}
58011026a{system_call+126}
80132bc6{schedule_tail+202}
580110d91{error_exit+0}
5 Badness in

RE: [asterisk-users] SIP Echo

I tried with the ping ... all of the phones respond in ca. 0.3ms, so network 
seems to be OK. More than 90% of CPU on * box is idle even in peak times, so 
this shouldn't cause echoes either, right? Hmmm, so handset could be an issue, 
but did anyone ever experience any handset problems with Polycom IP SoundPoint 
430 :-) ?

 

I will check the headsets and any possibilities of acoustical echo. Besides 
that, if we rule out the network, and the CPU on the * box, is there anything 
else that could be causing echoes on internal SIP calls?

 

  _  

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gomillion
Sent: Tuesday, May 22, 2007 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo

 

We experience echo too from time to time. It's usually headset-related, but not 
always. I ran a persistent ping on one of the phones, and we diagnosed a wiring 
problem with it. Other phones needed a new handset. The problem is that these 
problems need to be fixed on the phone NOT hearing echo. 

On 5/22/07, Asterisk [EMAIL PROTECTED] wrote:

How could I check if bandwith or/and latency is causing it?

If I do SIP show peers it says OK (13 ms) for all peers. I guess there is a way 
to gather more detailed info on SIP calls and latency?

* box is connected to the 1Gb switch with 1Gb connection, and clients have 
100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP hardphones 
connected to the * box. 

Thanks, Alex

-Original Message-
From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] 
On Behalf Of Alexandre VERNIOL
Sent: Tuesday, May 22, 2007 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo 

Hi,

Could be bandwith or/and latency ... Many causes...


Alex

Asterisk a écrit :
 Hello all,

 One of our clients reported that they are experiencing echo in SIP calls
 (even on internal ones). What do you think could be causing echo in
 internal SIP calls?

 We're using Polycom telephones, do you think they could be causing it?

 Thanks,
 Alex 

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[asterisk-users] Re: OK to have Asterisk and clients behind firewalls?

2007-05-22 Thread Vincent

On Tue, 22 May 2007 11:45:34 +0200, randulo [EMAIL PROTECTED]
wrote:
Upon replacement of the Linksys, everything worked fine except audio
on the Sipura. Turns out you need Symmetric RTP turned on in the phone
as Chris Mason says below.

Thanks for the tip. The IP phone doesn't have a setting that says
Symetric RTP, and turning it off in the 3102 doesn't make a
difference.

Here's the updated diagram:

http://codecomplete.free.fr/3102_nat/voip_no_sound_open_ports.jpg

I set up a syslog-ng server on the Asterisk host. I'm (obviously) no
SIP expert, but, although the remote IP phone's public IP comes up
twice towards the end, it looks like the Linksys sends RTP packets
Asterisk instead:

http://codecomplete.free.fr/3102_nat/calling_into_linksys.txt

Must be one of those problems that are solved in 2 seconds with the
right click or line in a configuration file... when you know what
you're doing :-)

Thank you.
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Re: [asterisk-users] SIP Echo

2007-05-22 Thread David Gomillion


Are your phones reinviting? Do you have any strange routing weirdness, or
are they all on a single subnet?

On 5/22/07, Asterisk [EMAIL PROTECTED] wrote:


 I tried with the ping ... all of the phones respond in ca. 0.3ms, so
network seems to be OK. More than 90% of CPU on * box is idle even in peak
times, so this shouldn't cause echoes either, right? Hmmm, so handset could
be an issue, but did anyone ever experience any handset problems with
Polycom IP SoundPoint 430 :-) ?



I will check the headsets and any possibilities of acoustical echo.
Besides that, if we rule out the network, and the CPU on the * box, is there
anything else that could be causing echoes on internal SIP calls?


 --

*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *David Gomillion
*Sent:* Tuesday, May 22, 2007 3:22 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] SIP  Echo



We experience echo too from time to time. It's usually headset-related,
but not always. I ran a persistent ping on one of the phones, and we
diagnosed a wiring problem with it. Other phones needed a new handset. The
problem is that these problems need to be fixed on the phone NOT hearing
echo.

On 5/22/07, *Asterisk* [EMAIL PROTECTED] wrote:

How could I check if bandwith or/and latency is causing it?

If I do SIP show peers it says OK (13 ms) for all peers. I guess there is
a way to gather more detailed info on SIP calls and latency?

* box is connected to the 1Gb switch with 1Gb connection, and clients have
100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP hardphones
connected to the * box.

Thanks, Alex

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Alexandre VERNIOL
Sent: Tuesday, May 22, 2007 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo

Hi,

Could be bandwith or/and latency ... Many causes...


Alex

Asterisk a écrit :
 Hello all,

 One of our clients reported that they are experiencing echo in SIP calls
 (even on internal ones). What do you think could be causing echo in
 internal SIP calls?

 We're using Polycom telephones, do you think they could be causing it?

 Thanks,
 Alex

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RE: [asterisk-users] Blackberry 8800+VoIP Configuration

2007-05-22 Thread Alexander Lopez

The only way I have ever seen any SIP and/or Network configurations is
from the Enterprise server management screen.  If you purchased the 8800
thru a participating carrier, RIM offers a single user express license
for free (with purchase) Google for Free BlackBerry Express and that
should give you the URL you need.

 

Alex

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Tuesday, May 22, 2007 5:30 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Blackberry 8800+VoIP Configuration

 

Hi Friends,

I bought a new Black Berry 8800 mobile. This mobile is coming with Wi-Fi
feature. After turning on this Wi-Fi feature in my mobile, It is not
detecting my wireless router in our office. How can I do this?

How can I configure my VoIP (SIP) server in this Blackberry 8800 mobile?

I tried a lot to do the above things in my mobile. But, I failed.

Look forward to your response. Thank you.

Regards,
Chandra.

  



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Nw--%0d%0a  to find flight and hotel bargains.

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Re: [asterisk-users] Local SMS how-to.

2007-05-22 Thread Jonson Player


Thank you for reply. Can you send me some working configs? I'm still
confusing about this sms option.

On 5/22/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:


Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player:
 Hello,
 i just want to activate SMS service between my asterisk local sip
 accounts and between asterisk and local sip accounts. How can i do
 this thin? Also i tried smsq to an account but all i obtained is a
 error message:

 ---Cut Here---
 May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to
 open /var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1:
 Permission denied, deleting
 May 22 13:09:37 WARNING[4829] pbx_spool.c: Failed to scan service
 '/var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1'
 ---And Here---

 Is necessary supplementary settings in /etc/asterisk/extensions.conf
 and /etc/asterisk/sip.conf ? Is necessary special module? I checked
 apps_sms.so is already loaded.

 Thank you for your support guys.

The SMSq stuff is for landline-type SMS, like those that never became
really popular here in Europe ;-) I do not know of any SIP hardphone
that supports them, but regular analog and ISDN handsets behind a
SIP-to-analog/ISDN gateway work for me.

The point of this SMS transfer method is calling the destination handset
with a certain callerid set (which differs between countries - whatever
number the telco prefers to choose - this can also be configured in the
phone). The phone will not ring but instead immediately answer the call
and receive the short message at 1200bps whatever modem standard they
chose to use.

For sending SMS, the handset will call a similarly telco-provided number
(premium-rate numbers here in Germany - maybe that is the reason for the
lack of popularity of this service) and do that 1200bps talk.

If you still think you can make use of it, make sure to call smsq with
the user id that asterisk is running as. That _might_ already do the
trick. If you do not get it running, ask again - I might have a working
setup somewhere around ;-)

Nevertheless, for me, landline SMS is a PITA. The only great thing is
you can upload Ringtones to Siemens gigaset phones.

BR
Anselm

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RE: [asterisk-users] SIP Echo

In Sip.conf I have the following: canreinvite=no 

 

No, all telephones are on the same subnet, handled by the same switch. I cannot 
verify if anything has been changed since I installed  configured the network, 
but as far as I know the whole network configuration is pretty straightforward, 
without any routing madness.

 

Kind Regards,

Alex

 

  _  

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gomillion
Sent: Tuesday, May 22, 2007 4:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo

 

Are your phones reinviting? Do you have any strange routing weirdness, or are 
they all on a single subnet?

On 5/22/07, Asterisk  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  wrote:

I tried with the ping ... all of the phones respond in ca. 0.3ms, so network 
seems to be OK. More than 90% of CPU on * box is idle even in peak times, so 
this shouldn't cause echoes either, right? Hmmm, so handset could be an issue, 
but did anyone ever experience any handset problems with Polycom IP SoundPoint 
430 :-) ?

 

I will check the headsets and any possibilities of acoustical echo. Besides 
that, if we rule out the network, and the CPU on the * box, is there anything 
else that could be causing echoes on internal SIP calls?

 

  _  

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gomillion
Sent: Tuesday, May 22, 2007 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo

 

We experience echo too from time to time. It's usually headset-related, but not 
always. I ran a persistent ping on one of the phones, and we diagnosed a wiring 
problem with it. Other phones needed a new handset. The problem is that these 
problems need to be fixed on the phone NOT hearing echo. 

On 5/22/07, Asterisk [EMAIL PROTECTED] wrote:

How could I check if bandwith or/and latency is causing it?

If I do SIP show peers it says OK (13 ms) for all peers. I guess there is a way 
to gather more detailed info on SIP calls and latency?

* box is connected to the 1Gb switch with 1Gb connection, and clients have 
100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP hardphones 
connected to the * box. 

Thanks, Alex

-Original Message-
From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] 
On Behalf Of Alexandre VERNIOL
Sent: Tuesday, May 22, 2007 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo 

Hi,

Could be bandwith or/and latency ... Many causes...


Alex

Asterisk a écrit :
 Hello all,

 One of our clients reported that they are experiencing echo in SIP calls
 (even on internal ones). What do you think could be causing echo in
 internal SIP calls?

 We're using Polycom telephones, do you think they could be causing it?

 Thanks,
 Alex 

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Re: [asterisk-users] Why 2 branches of asterisk development?

2007-05-22 Thread Tony Plack

Because there is a huge install base on 1.2 which is fairly stable but still needs bug fix/security patches. There are no new features being developed by the code group on this version, but there are outside people who are still modifying it.

1.4 has many new features and some areas re-written. This makes the code base less stable but usable. The feature set is complete for this therefore, it was released. Now it is up to us bold souls to verify it works. At some time, when the 1.2 base starts to wane, I would see the 1.2 going away, just not now.

1.6 is where new development/features are being planed. I hope they finally write the SIP stack so it lines up with the SIP specification instead of being a adjunct to the IAX code. I am not looking for SER type functionality but better performance and support for things like CallInfo messages being passed from Origin to Destination.

On the whole, most proprietary software companies support their older versions. Microsoft has patches being developed for versions back to Windows 2000.
Hi all,
i never understood that why is there 2 branches of asterisk going
on parallel. asterisk 1.2.* and asterisk 1.4.*, i also heard about
beginning of another branch which will be 1.6.*. so whats the
difference between these 2 or 3 versions, can anybody plz tel me?

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Re: [asterisk-users] SIP Echo

2007-05-22 Thread Anthony Francis


Asterisk wrote:


In Sip.conf I have the following: canreinvite=no

 

No, all telephones are on the same subnet, handled by the same switch. 
I cannot verify if anything has been changed since I installed  
configured the network, but as far as I know the whole network 
configuration is pretty straightforward, without any routing madness.


 


Kind Regards,

Alex

 




*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *David 
Gomillion

*Sent:* Tuesday, May 22, 2007 4:34 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] SIP  Echo

 

Are your phones reinviting? Do you have any strange routing weirdness, 
or are they all on a single subnet?


On 5/22/07, *Asterisk*  [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


I tried with the ping ... all of the phones respond in ca. 0.3ms, so 
network seems to be OK. More than 90% of CPU on * box is idle even in 
peak times, so this shouldn't cause echoes either, right? Hmmm, so 
handset could be an issue, but did anyone ever experience any handset 
problems with Polycom IP SoundPoint 430 :-) ?


 

I will check the headsets and any possibilities of acoustical echo. 
Besides that, if we rule out the network, and the CPU on the * box, is 
there anything else that could be causing echoes on internal SIP calls?


 




*From:* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]] *On Behalf Of *David 
Gomillion

*Sent:* Tuesday, May 22, 2007 3:22 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] SIP  Echo

 

We experience echo too from time to time. It's usually 
headset-related, but not always. I ran a persistent ping on one of the 
phones, and we diagnosed a wiring problem with it. Other phones needed 
a new handset. The problem is that these problems need to be fixed on 
the phone NOT hearing echo.


On 5/22/07, *Asterisk* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


How could I check if bandwith or/and latency is causing it?

If I do SIP show peers it says OK (13 ms) for all peers. I guess there 
is a way to gather more detailed info on SIP calls and latency?


* box is connected to the 1Gb switch with 1Gb connection, and clients 
have 100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP 
hardphones connected to the * box.


Thanks, Alex

-Original Message-
From: [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] [mailto: 
[EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]] On Behalf Of 
Alexandre VERNIOL

Sent: Tuesday, May 22, 2007 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo

Hi,

Could be bandwith or/and latency ... Many causes...


Alex

Asterisk a écrit :
 Hello all,

 One of our clients reported that they are experiencing echo in SIP calls
 (even on internal ones). What do you think could be causing echo in
 internal SIP calls?

 We're using Polycom telephones, do you think they could be causing it?

 Thanks,
 Alex

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watch the comm between the switch and a host with issues using tcpdump 
-Avvv host host-ip


This can give you an idea of what is going on, also if they are all in 
the same subnet, you are giving your ast box unnecessary strain forcing 
it to relay the RTSP stream, I would set that canreinvite to yes. For 
reference on sip and how re-invite works, please read

[asterisk-users] Re: Kernel Panic in wct4xxp during unload on Zaptel-1.4.4

2007-05-22 Thread Axel Thimm

On Tue, May 22, 2007 at 10:07:19AM -0400, James FitzGibbon wrote:
 I attempted an upgrade of our production system from Asterisk/Zaptel 1.2 to
 1.4 this weekend.  Intially everything looked like it was working properly,
 but some time in the day following the upgrade, the system died to a kernel
 panic.  I wasn't able to catch the entire kernel dump on the console
 unfortunately.
 
 I attempted to isolate the panic, and found that when 'service zaptel stop'
 was run (specifically, when wct4xxp was unloaded) I get this panic
 consistently:
 
 5 Not prepped yet! (repeated approx 250 times)
 5 Freed a Wildcard
 5 Not prepped yet! (repeated approx 550 times)
 5 Stopped TE4XXP, Turned off DMA
 5 Not prepped yet! (repeated approx 11000 times)
 5 Unable to handle kernel paging request at ff034010 RIP:
 5 a0163207{:wct4xxp:t4_interrupt_gen2+63}
 5 PML4 4ea063 PGD 1388067 PMD 1389067 PTE 0
 5 Oops:  [1] SMP
 5 CPU 1
 5 Modules linked in: zttranscode(U) wct4xxp(U) zaptel(U) crc_ccitt
 netconsole md5 ipv6 dm_mirror dm_mod button battery ac joydev ehci_hcd
 uhci_hcd hw_random bnx2 ext3 jbd
 cciss sd_mod scsi_mod
 5 Pid: 11053, comm: hotplug Not tainted 2.6.9-42.0.8.ELsmp
 5 RIP: 0010:[a0163207]
 a0163207{:wct4xxp:t4_interrupt_gen2+63}
 5 RSP: :0100013ebdb0  EFLAGS: 00010046
 5 RAX: ff034000 RBX: 010073478724 RCX: 0002
 5 RDX: 010073478680 RSI: 0002 RDI: 010073478724
 5 RBP: 010073478680 R08: 0008 R09: 
 5 R10:  R11: 0002 R12: 00d1
 5 R13: 0100013ebec8 R14: 0100013ebec8 R15: 010075e94978
 5 FS:  002a955643e0() GS:804e5900()
 knlGS:
 5 CS:  0010 DS:  ES:  CR0: 8005003b
 5 CR2: ff034010 CR3: 013d8000 CR4: 06e0
 5 Process hotplug (pid: 11053, threadinfo 01007413c000, task
 01007c445030)
 5 Stack: 00d1 01007413dc98 80138552
 0038
 50100013ebea8 0100013ebde8 0001
 00d1
 50012 010073478680
 5 Call Trace:IRQ 80138552{printk+141}
 80112f4a{handle_IRQ_event+41}
 5801131c4{do_IRQ+197}
 80110833{ret_from_intr+0}
 58013c731{__do_softirq+77}
 8013c7e5{do_softirq+49}
 580110bf5{apic_timer_interrupt+133}  EOI
 8011c21a{flush_tlb_page+44}
 580169106{do_wp_page+1127}
 80123ed3{do_page_fault+575}
 580169ff2{handle_mm_fault+1228}
 80123e9a{do_page_fault+518}
 58011026a{system_call+126}
 80132bc6{schedule_tail+202}
 580110d91{error_exit+0}
 5
 5 Code: 8b 40 10 89 44 24 58 e8 3d 80 1a e0 31 c0 f6 44 24 58 07 0f
 5 RIP a0163207{:wct4xxp:t4_interrupt_gen2+63} RSP
 0100013ebdb0
 5 CR2: ff034010
 5  0Kernel panic - not syncing: Oops
 5  Badness in panic at kernel/panic.c:118
 5
 5 Call Trace:IRQ 80137a8a{panic+527}
 80110bf5{apic_timer_interrupt+133}
 580111aec{oops_end+38} 80111b07{oops_end+65}
 580124148{do_page_fault+1204}
 a0078f51{:bnx2:bnx2_start_xmit+470}
 5802bb4cd{netpoll_send_skb+257}
 80110d91{error_exit+0}
 5a0163207{:wct4xxp:t4_interrupt_gen2+63}
 80138552{printk+141}
 580112f4a{handle_IRQ_event+41}
 801131c4{do_IRQ+197}
 580110833{ret_from_intr+0}
 8013c731{__do_softirq+77}
 58013c7e5{do_softirq+49}
 80110bf5{apic_timer_interrupt+133}
 5 EOI 8011c21a{flush_tlb_page+44}
 80169106{do_wp_page+1127}
 580123ed3{do_page_fault+575}
 80169ff2{handle_mm_fault+1228}
 580123e9a{do_page_fault+518}
 8011026a{system_call+126}
 580132bc6{schedule_tail+202}
 80110d91{error_exit+0}
 5
 5 Badness in i8042_panic_blink at drivers/input/serio/i8042.c:987
 5
 5 Call Trace:IRQ 8024219b{i8042_panic_blink+238}
 80137a38{panic+445}
 580110bf5{apic_timer_interrupt+133}
 80111aec{oops_end+38}
 580111b07{oops_end+65}
 80124148{do_page_fault+1204}
 5a0078f51{:bnx2:bnx2_start_xmit+470}
 802bb4cd{netpoll_send_skb+257}
 580110d91{error_exit+0}
 a0163207{:wct4xxp:t4_interrupt_gen2+63}
 580138552{printk+141}
 80112f4a{handle_IRQ_event+41}
 5801131c4{do_IRQ+197}
 80110833{ret_from_intr+0}
 58013c731{__do_softirq+77}
 8013c7e5{do_softirq+49}
 580110bf5{apic_timer_interrupt+133}  EOI
 8011c21a{flush_tlb_page+44}
 580169106{do_wp_page+1127}
 80123ed3{do_page_fault+575}
 580169ff2{handle_mm_fault+1228}

Re: [asterisk-users] how can I catch the event generated when a parked call is hung up?


[EMAIL PROTECTED] wrote:

Hi all,

how can I catch the event generated when a parked call is hung up?

In my dialplan, when arrives a call to a specific number, Asterisk parks
the call and announces the parking slot to a number. But if the user hangs
up the parked call, I don't know how to catch the event, from dialplan,
that removes the call from the parking slot. I want to know if there is a
method for do this.

Thank you!


That's a good question actually.  Have you tried adding the h 
extension to the context where the call is parked?


exten=123,1,Dial(SIP/123,30,m)   ; == park when answered here
exten=h,1,Noop(Damn it.  That one got a away!)

I wonder if that would work.  No time to try it myself...

--

Warm Regards,

Lee



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Re: [asterisk-users] SIP Echo

2007-05-22 Thread Jorge Mendoza


Try  canreinvite=yes in order to confirm that CPU is not the problem.

Jorge Mendoza

Asterisk wrote:


I tried with the ping ... all of the phones respond in ca. 0.3ms, so 
network seems to be OK. More than 90% of CPU on * box is idle even in 
peak times, so this shouldn't cause echoes either, right? Hmmm, so 
handset could be an issue, but did anyone ever experience any handset 
problems with Polycom IP SoundPoint 430 :-) ?


 

I will check the headsets and any possibilities of acoustical echo. 
Besides that, if we rule out the network, and the CPU on the * box, is 
there anything else that could be causing echoes on internal SIP calls?


 




*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *David 
Gomillion

*Sent:* Tuesday, May 22, 2007 3:22 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] SIP  Echo

 

We experience echo too from time to time. It's usually 
headset-related, but not always. I ran a persistent ping on one of the 
phones, and we diagnosed a wiring problem with it. Other phones needed 
a new handset. The problem is that these problems need to be fixed on 
the phone NOT hearing echo.


On 5/22/07, *Asterisk* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


How could I check if bandwith or/and latency is causing it?

If I do SIP show peers it says OK (13 ms) for all peers. I guess there 
is a way to gather more detailed info on SIP calls and latency?


* box is connected to the 1Gb switch with 1Gb connection, and clients 
have 100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP 
hardphones connected to the * box.


Thanks, Alex

-Original Message-
From: [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] [mailto: 
[EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]] On Behalf Of 
Alexandre VERNIOL

Sent: Tuesday, May 22, 2007 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo

Hi,

Could be bandwith or/and latency ... Many causes...


Alex

Asterisk a écrit :
 Hello all,

 One of our clients reported that they are experiencing echo in SIP calls
 (even on internal ones). What do you think could be causing echo in
 internal SIP calls?

 We're using Polycom telephones, do you think they could be causing it?

 Thanks,
 Alex

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RE: [asterisk-users] SIP Echo

2007-05-22 Thread Dan Austin

Alex wrote:
 I tried with the ping ... all of the phones respond 
 in ca. 0.3ms, so network seems to be OK. More than 
 90% of CPU on * box is idle even in peak times, so 
 this shouldn't cause echoes either, right? Hmmm, so 
 handset could be an issue, but did anyone ever 
 experience any handset problems with Polycom IP 
 SoundPoint 430 :-) ?

 I will check the headsets and any possibilities of 
 acoustical echo. Besides that, if we rule out the 
 network, and the CPU on the * box, is there anything 
 else that could be causing echoes on internal SIP calls?

As others have pointed out it is highly unlikely that
a network issue is the source of the problem (unless
the phone's firmware has a MAJOR bug).

Acoustic issue is 99.9% likely to be the cause, but
is can be less than obvious why.  A certain vintage
of Cisco phone firmware would introduce echo when the
headset/handset/speaker volume was set above 65~75%.

I spent about 6 months chasing that one on and off.
After Cisco fixed that, then next two common causes
were:
   1.  Enclosed offices/conference rooms without
acoustic treatment 
   2.  3rd party amplified headsets
  (Echo was only one symptom of this one and not
a common one, but it did happen)

Some phones deal with item 1 better than others.
last ditch efforts to fixup a room that a phone
has problems with would include wall hangings, or
even a cloth place mat (don't use the wife's Holiday
mats) under the phone if the echo is most common on
speakerphone calls.  I've often wondered why phone
designers put the mic on the bottom front of so many
phones, where it is most likely to get acoustic reflection
off the table/desk surface...

Oh, one more cause that is a bear to correct.  After
first switching to the new system, my users felt the
need to yell at their phones.  Maybe a byproduct of 
poor experience with cell phones, which is how they
expected the new phones to work like.  Getting the
yellers and loud talkers to bring it down a notch also
helped.

Dan
 
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RE: [asterisk-users] SIP Echo

Thanks guys for the tips. I will try that.

Alex

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis
Sent: Tuesday, May 22, 2007 5:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP  Echo

Asterisk wrote:

 In Sip.conf I have the following: canreinvite=no

 No, all telephones are on the same subnet, handled by the same switch. 
 I cannot verify if anything has been changed since I installed  
 configured the network, but as far as I know the whole network 
 configuration is pretty straightforward, without any routing madness.

 Kind Regards,

 Alex

 *From:* [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] *On Behalf Of *David 
 Gomillion
 *Sent:* Tuesday, May 22, 2007 4:34 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] SIP  Echo

 Are your phones reinviting? Do you have any strange routing weirdness, 
 or are they all on a single subnet?

 On 5/22/07, *Asterisk*  [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] wrote:

 I tried with the ping ... all of the phones respond in ca. 0.3ms, so 
 network seems to be OK. More than 90% of CPU on * box is idle even in 
 peak times, so this shouldn't cause echoes either, right? Hmmm, so 
 handset could be an issue, but did anyone ever experience any handset 
 problems with Polycom IP SoundPoint 430 :-) ?

 I will check the headsets and any possibilities of acoustical echo. 
 Besides that, if we rule out the network, and the CPU on the * box, is 
 there anything else that could be causing echoes on internal SIP calls?

 *From:* [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED]] *On Behalf Of *David 
 Gomillion
 *Sent:* Tuesday, May 22, 2007 3:22 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] SIP  Echo

 We experience echo too from time to time. It's usually 
 headset-related, but not always. I ran a persistent ping on one of the 
 phones, and we diagnosed a wiring problem with it. Other phones needed 
 a new handset. The problem is that these problems need to be fixed on 
 the phone NOT hearing echo.

 On 5/22/07, *Asterisk* [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] wrote:

 How could I check if bandwith or/and latency is causing it?

 If I do SIP show peers it says OK (13 ms) for all peers. I guess there 
 is a way to gather more detailed info on SIP calls and latency?

 * box is connected to the 1Gb switch with 1Gb connection, and clients 
 have 100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP 
 hardphones connected to the * box.

 Thanks, Alex

 -Original Message-
 From: [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] [mailto: 
 [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED]] On Behalf Of 
 Alexandre VERNIOL
 Sent: Tuesday, May 22, 2007 2:18 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] SIP  Echo

 Hi,

 Could be bandwith or/and latency ... Many causes...

 Alex

 Asterisk a écrit :
  Hello all,

  One of our clients reported that they are experiencing echo in SIP calls
  (even on internal ones). What do you think could be causing echo in
  internal SIP calls?

  We're using Polycom telephones, do you think they could be causing it?

  Thanks,
  Alex

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watch the

Re: [asterisk-users] how can I catch the event generated when a parked call is hung up?

2007-05-22 Thread Anthony Francis


Lee Jenkins wrote:

[EMAIL PROTECTED] wrote:

Hi all,

how can I catch the event generated when a parked call is hung up?

In my dialplan, when arrives a call to a specific number, Asterisk parks
the call and announces the parking slot to a number. But if the user 
hangs

up the parked call, I don't know how to catch the event, from dialplan,
that removes the call from the parking slot. I want to know if there 
is a

method for do this.

Thank you!


That's a good question actually.  Have you tried adding the h 
extension to the context where the call is parked?


exten=123,1,Dial(SIP/123,30,m)   ; == park when answered here
exten=h,1,Noop(Damn it.  That one got a away!)

I wonder if that would work.  No time to try it myself...

if you are monitoring through AMI userevent is better than NoOp because 
you can create a custom event to trap against.

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[asterisk-users] Re: how can I catch the event generated when a parked call is hung up?

2007-05-22 Thread lavarini

This method does not seem to work. The action (NoOp in my case) in the h
extension is execute after have parked the call, while when I hang up the
call parked the action in h extension is not execute.


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[asterisk-users] Net 2 Phone - Asterisk - Problem

2007-05-22 Thread Milton Davila


Hi there,

I am having some problems while trying to place phone calls through Asterisk
to Net2phone, this is my setup:

I have a SIP phone connected directly to my Asterisk box from where I want
the call to origin; in sip.conf:

[mySIP]
type=friend
username=mySIP
secret=mySecret
host=dynamic
context=outgoing

I read that I have to make some changes in sip.conf, in order to make it
work with Net2phone:

http://www.voip-info.org/wiki/view/Asterisk+settings+Net2phone

So these are the changes I made in sip.conf:

[general]
useragent = X-Lite release 1103m
register = PHONENUMBER:[EMAIL PROTECTED]

[net2phone]
type = peer
host = sip.net2phone.com
username = PHONENUMBER
secret = PASSWORD
fromuser = PHONENUMBER
fromdomain = net2phone.com
context = incoming
insecure = very
canreinvite = no

Now here is my extensions.conf:

[outgoing]
exten = _9NXXNXX,1,Dial(SIP/net2phone/${EXTEN:1})

If I type sip show registry in the Asterisk console, it shows that the
state of the Net2phone sip is Registered.

The problem is that when I call any phone is USA: 1-XXX-XXX-, I only get
a busy tone. So I can never really place a call.

What can be the problem?

I am using Asterisk 1.4.2 on Red Hat Enterprise Linux 5.
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Re: [asterisk-users] SIP Echo

2007-05-22 Thread Stephen Bosch

Asterisk wrote:
 I tried with the ping ... all of the phones respond in ca. 0.3ms, so
 network seems to be OK. More than 90% of CPU on * box is idle even in
 peak times, so this shouldn't cause echoes either, right? Hmmm, so
 handset could be an issue, but did anyone ever experience any handset
 problems with Polycom IP SoundPoint 430 :-) ?

Headsets are a terrible source of echo.

Are you using a headset amplifier? Polycom specifically recommends use
of an amplifier with the SoundPoint IP phones (most of the newer ones
have integrated echo cancellation).

-Stephen-

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Re: [asterisk-users] Re: how can I catch the event generated when a parked call is hung up?


[EMAIL PROTECTED] wrote:

This method does not seem to work. The action (NoOp in my case) in the h
extension is execute after have parked the call, while when I hang up the
call parked the action in h extension is not execute.



So Asterisk sees the parking of the call as the hanging up of that channel.

Maybe AMI is the only way to go then...sorry I couldn't be more help. 
The above was only a guess on my part that seemed to make sense.


--

Warm Regards,

Lee



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Re: [asterisk-users] SIP Echo

2007-05-22 Thread Crazy Boy

Hi,

Did you implement QoS (Quality of Service) in your network? 

Thanks.

Regards,
Chandra

Stephen Bosch [EMAIL PROTECTED] wrote: Asterisk wrote:
 I tried with the ping ... all of the phones respond in ca. 0.3ms, so
 network seems to be OK. More than 90% of CPU on * box is idle even in
 peak times, so this shouldn't cause echoes either, right? Hmmm, so
 handset could be an issue, but did anyone ever experience any handset
 problems with Polycom IP SoundPoint 430 :-) ?

Headsets are a terrible source of echo.

Are you using a headset amplifier? Polycom specifically recommends use
of an amplifier with the SoundPoint IP phones (most of the newer ones
have integrated echo cancellation).

-Stephen-

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-
Boardwalk for $500? In 2007? Ha! 
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Re: [asterisk-users] MoH WAY too loud

2007-05-22 Thread Andrew Kohlsmith

On Monday 21 May 2007 3:38 pm, Doug Lytle wrote:
 Doing a 'man sox' does wonders:

The question, however, is is Asterisk playing them louder than normal, or are 
they recorded too loudly to begin with?

Adjusting volume gains on these files is the LAST thing you should do.  
Determine what the nature of the problem is, precisely, before resorting to 
these hacks.

-A.
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[asterisk-users] Phones fail to ring

2007-05-22 Thread Jim Suber

I am somewhat confused. I have the incoming (s) context playing a greeting
and callers choose one of two extensions (100, or 101)

To the caller it ALWAYS sounds as though the phone is ringing. However,
sometimes it is not actually ringing the phones 

The listext.wav file suggests extensions 100 or 101

 

exten = s,1,Zapateller(nocallerid)

exten = s,2,Answer()

exten = s,3,Background(listext)

exten = i,1,PlayBack(pbx-invalid)

exten = i,2,Goto(incoming,s,1)

exten = t,1,PlayBack(vm-goodbye)

exten = t,2,Hangup()

 

Thanks in advance

Jim

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[asterisk-users] Astsee v0.5 now available, X/Linux Asterisk Usage Auditor and Monitor

2007-05-22 Thread Mojo with Horan Company, LLC

The latest and greatest Astsee is now available at 
http://www.astsee.com/ I'm up to v0.5 today -- the light at the end of 
the tunnel edition.


In progress is a way to audit this sort of traffic _without_ manager
credentials  ;)  Just by sniffing it off the wire or out of the air...

You can test that out now by running
./astsee any
instead of
./astsee host port user secret

but unless you are:
  a) using a hub instead of a switch or
  b) using a mirroring port of a cool switch or
  c) running something like arpspoof or
  d) something else?
you of course won't see anyone else's traffic  :(

but you can for example run it in sniff mode and then use another 
manager program or telnet alone to simulate some manager event traffic 
-- works like a charm  :)


Bugs, comments, suggestions welcome!

Mojo
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Re: [asterisk-users] Dry Copper Pair

2007-05-22 Thread Andrew Kohlsmith

On Sunday 20 May 2007 11:36 am, Jon Pounder wrote:
 how many cable feet were you ever able to actually get various speeds at ?

Depended on the hardware and wire gauge.  I was able to do 1250kbps 
symmetrical on a 4kmish loop very reliably.

 around here it might just be the geography but I think load coils are
 really just a well talked about myth. There are no truly long haul
 lines due to the number of cities so close together and the lakes
 blocking what would be any longer haul lines.

Load coils are no myth, at least in rural Ontario (Canada) -- I've had to have 
them removed on more than one occasion.

-A.
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[asterisk-users] SMS

2007-05-22 Thread Andre Courchesne - Consultant


Hi,

  Anyone has details or information on how to use the SMS command to send SMS 
to Fido, Bell Mobility and Rogers Wireless in Canada?


  Thanks,

Andre Courchesne
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Re: [asterisk-users] Local SMS how-to.

2007-05-22 Thread Yuan LIU

From: Anselm Martin Hoffmeister [EMAIL PROTECTED]
Date: Tue, 22 May 2007 13:41:43 +0200

Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player:
 Hello,
 i just want to activate SMS service between my asterisk local sip
 accounts and between asterisk and local sip accounts. How can i do
 this thin? Also i tried smsq to an account but all i obtained is a
 error message:

 ---Cut Here---
 May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to
 open /var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1:
 Permission denied, deleting
 May 22 13:09:37 WARNING[4829] pbx_spool.c: Failed to scan service
 '/var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1'
 ---And Here---

 Is necessary supplementary settings in /etc/asterisk/extensions.conf
 and /etc/asterisk/sip.conf ? Is necessary special module? I checked
 apps_sms.so is already loaded.

 Thank you for your support guys.

No special change in sip.conf required.  I've transmitted SMS over local SIP 
channel and it's be quire reliable - over LAN.

Yuan Liu

The SMSq stuff is for landline-type SMS, like those that never became
really popular here in Europe ;-) I do not know of any SIP hardphone
that supports them, but regular analog and ISDN handsets behind a
SIP-to-analog/ISDN gateway work for me.

The point of this SMS transfer method is calling the destination handset
with a certain callerid set (which differs between countries - whatever
number the telco prefers to choose - this can also be configured in the
phone). The phone will not ring but instead immediately answer the call
and receive the short message at 1200bps whatever modem standard they
chose to use.

For sending SMS, the handset will call a similarly telco-provided number
(premium-rate numbers here in Germany - maybe that is the reason for the
lack of popularity of this service) and do that 1200bps talk.

If you still think you can make use of it, make sure to call smsq with
the user id that asterisk is running as. That _might_ already do the
trick. If you do not get it running, ask again - I might have a working
setup somewhere around ;-)

Nevertheless, for me, landline SMS is a PITA. The only great thing is
you can upload Ringtones to Siemens gigaset phones.

BR
Anselm

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Re: [asterisk-users] Zaptel hangs machine...

2007-05-22 Thread Angel Luis Martinez


Tzafrir Cohen escribió:

On Sat, May 19, 2007 at 07:26:33PM +0200, Angel Luis Martinez wrote:
  

Hi all.

When i do a service zaptel stop on my machine,sometimes it crash and i
must unplug and plug the power cord to restart the machine. Also sometimes
load zttranscode and wct4xxp, and oter times wct4xxp only... it's running
centos 4.4, libpri 1.2.4, zaptel 1.2.17.1 and asterisk 1.2.18. with a
TE212P.

Can somebody help me?



Try editing /etc/init.d/zaptel and remove 'ztcfg -s' from there.

  

Thanks for you response. It works !!!

Thanks a lot !!
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[asterisk-users] how to disable global authentication for registration

2007-05-22 Thread Jason Ma


Buddies,
I am new guy here,I installed Asterisk 1.44 and setup AsteriskNow
manually.Iwant to disable the global digest authentication for
registration so that I
can easily to test my Asterisk system with another call generation tool,how
can I do that?Will appreciate for any replies.Thanks in advance!
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Re: [asterisk-users] Mobile Number to Mobile carrier mapping

2007-05-22 Thread Ritesh Agrawal

Hi Alex,
This is a nice summary. Thanks a lot for your response.
My mere interest was to find out
(1) if a number is a mobile number
(2) If #1 is true, then if I had the carrier name, I could generate an SMS
to the US phone number without asking for the carrier info.

Ritesh

On 5/19/07, Alex Balashov [EMAIL PROTECTED] wrote:

On Wed, 9 May 2007, Ritesh Agrawal said something to this effect:

Is there a way to find out the mobile/landline carrier name based on the
phone number?

Ordinary people can only find this out if the NPA-NXX (area code +
exchange, i.e. the first six digits) block to which the number belongs
is assigned or delegated to a particular mobile carrier. So, what you'd
really be looking up is a particular NPA-NXX block's registered ownership.

There are many ways to get this information. You can go to
localcallingguide.com and do an Area Code/Prefix/OCN search. There's
also telcodata.us, and I imagine some others. Or you can download the
NXX block assignment spreadsheet straight from NANPA's web site. This
type of CO information is public and relatively ubiquitous, if you know
where to look.

One caveat is that this information can be somewhat out of date or
inaccurate, especially in 1-blocks that have subdelegations across
carriers.

The other is that this will not properly identify a phone number's
origin
for you if it's been ported away from the block-owning carrier under the
Local Number Portability regime, to someone else in the LATA. This trend
has become especially accelerated with the advent of VoIP, when there is
additional incentive to get your service from another LEC because it's not
just purely a matter of someone's POTS vs. someone else's POTS (or ISDN or
whatever).

To really know what OCN (Operating Carrier Number) a number is assigned
for sure, you have to make a query against Neustar's NPAC database, which
SS7 STPs use to do LNP dips. Most mere mortals do not have that ability
readily at their disposal, as for the most part any kind of visibility
into
NPAC is contingent upon being a carrier and operating a switch. Some
service providers that are not carriers may have it as well, and I don't
really know what Neustar's guidelines for that are.

Based on localcallingguide.com, the number you provided is a
CommPartners
number, as per:

http://www.localcallingguide.com/lca_prefix.php?npa=415nxx=234x=ocn=region=lata=switch=pastdays=0nextdays=0

An LNP dip confirms that this number is in fact part of CommPartners,
but
shows it is not in that original OCN. It is under OCN 533C, which is also
CommPartners, but possibly a slightly different trunking handoff, or
whatever the logistical difference is.

Hope that helps,

-- Alex

--
Alex Balashov [EMAIL PROTECTED]
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[asterisk-users] Working softphone for poket PC

2007-05-22 Thread Cosmin Prund

Googling arround I found a number of pocket pc softphones. Of those I was only 
able to install SJ-something (removed it).

Is there one (pocket pc softphone) that works?

Thanks,
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Re: [asterisk-users] Local SMS how-to.

Am Dienstag, den 22.05.2007, 17:35 +0300 schrieb Jonson Player:
 Thank you for reply. Can you send me some working configs? I'm still
 confusing about this sms option.

Just to get you started, try this:

Find out which user asterisk runs as. Get a shell for that user.
Run (all in one line)

smsq --mt --oa=321 --mttx-callerid=01930101 --mttx-channel=SIP/abcde
message text goes here

where 321 will displayed as sender id on the handset, and 01930101
will have to replaced by the mobile center known to your phone, plus 1
at the end - the German T-Com seems to use 0193010, and this setting
works for me. Further, SIP/abcde must be the channel that a SMS-capable
handset is available on: If you have some ATA with a DECT handset
connected, or similar, use the channel name exactly as you would in the
Dial() command.

First thing to find out is if this works. Be sure to have asterisk in
extra-verbose running a console to see what happens.

If the mobile handset rings (instead of getting the SMS) either the
01930101 number has not been set correctly or it probably is not
compatible with Asterisk SMS.

Once you get this far, you would need the other way round. When your
mobile phone tries to _send_ a text message, it will go to 01930100 (sms
center number plus 0). You will have to care for that in your
extensions.conf, like this

exten = 01930100,1,Wait(2)
exten = 01930100,2,Answer()
exten = 01930100,3,Wait(2)
exten = 01930100,4,SMS(01930100,as)
exten = 01930100,5,Wait(2)
exten = 01930100,6,Hangup()

In my experience those Wait(2) improve reliability over internet
connections, they probably are superfluous if you have reliable
low-latency LAN. For me, they made the difference between 10/100 and
95/100 successfuly sent messages.

You will have to write your own scriptwork to play with the files that
will be created from those commands. Their structure is simple, you will
find out.

Sending EMS (for ringtones and bitmaps) is a bit more complex, you will
need the UDH flag for that. I think I documented that once on this ML
but am not sure. However, it is possible with some Siemens Gigaset
devices, and pictures or monophonic ringtones.

BR
Anselm

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Re: [asterisk-users] Working softphone for poket PC

2007-05-22 Thread VoIP User


ppciax works too. And it is IAX2 softphone. Anyway, SJPhone is much better.

On 5/22/07, Cosmin Prund [EMAIL PROTECTED] wrote:


Googling arround I found a number of pocket pc softphones. Of those I was
only able to install SJ-something (removed it).

Is there one (pocket pc softphone) that works?

Thanks,
Cosmin Prund___
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Re: [asterisk-users] Working softphone for poket PC

Am Dienstag, den 22.05.2007, 21:49 +0300 schrieb Cosmin Prund:
 Googling arround I found a number of pocket pc softphones. Of those I was 
 only able to install SJ-something (removed it).
 
 Is there one (pocket pc softphone) that works?

When I searched for one, about half a year ago, there were two that
actually worked, but both had their flaws. One was SJphone, and that was
hard to get running. The other one was a Microsoft thingy, from their
developers ressources or whatever, that always used the loudspeaker
instead of the earphone piece...

Somehow they worked, but back then, I decided against and got a separate
WLAN phone from ebay. Not that that turned out to work more reliably,
mind, but at least some more men's toys ;-)

I would be glad to learn about a Wince softphone that actually worked
without choking on something like a phonenumber callerid starting +,
or just the random PDA crash that makes the reset button wear out.

Best,
Anselm

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Re: [asterisk-users] Mobile Number to Mobile carrier mapping

2007-05-22 Thread Alex Balashov


On Tue, 22 May 2007, Ritesh Agrawal said something to this effect:


Hi Alex,
This is a nice summary. Thanks a lot for your response.
My mere interest was to find out
(1) if a number is a mobile number
(2) If #1 is true, then if I had the carrier name, I could generate an SMS
to the US phone number without asking for the carrier info.


  In that case, using one of the databases I mentioned should work.  You 
probably can't deploy a real-time service based on dynamic queries to

LocalCallingGuide.com (even though they do have an XML query interface)
without working out some sort of arrangement with them, and another
option is the LERG.

  Of course, when I say work, I mean you'll be right about 98% of
the time.  The other times, the number's ported, and you won't know.
The only true way to do this is to use an LNP dip via SS7 or some API
to Neustar's NPAC that I don't know about.

-- Alex

--
Alex Balashov   [EMAIL PROTECTED]
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Re: [asterisk-users] Working softphone for poket PC

2007-05-22 Thread Remco Post

Cosmin Prund wrote:
 Googling arround I found a number of pocket pc softphones. Of those I
 was only able to install SJ-something (removed it).
 

SJphone, and why did you remove it?

 Is there one (pocket pc softphone) that works?
 

SJphone ;-) At least I've made some successful calls using sjphone

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-- 

Remco Post

I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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[asterisk-users] FW: autologoff

2007-05-22 Thread Ed Nuñez



Is the autologoff function supported in Asterisk BE B.1-3?  I have
configured my agents.conf with a 5 second timeout, but the agents extension
continues ringing until the call eventually goes to voicemail.



Agents.conf
[general]
persistentagents=yes


[agents]
autologoff = 5
multiplelogin = no
recordagencalls = yes
monitor-join = yes
createlink = yes
updatecdr = yes
musiconhold = default
recordformat = wav49
savecallsin = /var/spool/asterisk/monitor/

agent = 1650,1650,Tareq 
agent = 1656,1656,Ed 
agent = 2000,2000,test agent
agent = 1704,1704,Reload Test



queues.conf
[general]
persistentmembers=yes


[noi-cust-serv-spanish]
strategy = leastrecent
announce-frequency = 30
announce-holdtime = yes
announce-round-seconds = 10
timeout=180
monitor-format=wav49
monitor-join=yes
joinempty = strict
leavewhenempty = strict
musiconhold = default
eventwhencalled = yes
servicelevel=180
reportholdtime =yes
maxlen=0; maximum ammount of calls waiting
queue-youarenext = queue-youarenext;   (You are now first
in line.)
queue-thereare = queue-thereare;   (There are)
queue-callswaiting = queue-callswaiting;   (calls waiting.)
queue-holdtime = queue-holdtime;   (The current est.
holdtime is)
queue-minutes = queue-minutes  ;   (minutes.)
queue-seconds = queue-seconds  ;   (seconds.)
queue-thankyou = queue-thankyou;   (Thank you for your
patience.)
queue-lessthan = queue-less-than   ;   (less than)
queue-reporthold = queue-reporthold

member = Agent/1656


autologoff - with this option you set for how long the phone has to ring
with no answer, before the agent to be logged off. You have to set the
maximum period of time in seconds. By default this option is set to 15
seconds.


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[asterisk-users] Issue installing TE212P -- Echo Cancellor not working -- VPM450: Not Present

2007-05-22 Thread Deepak Naidu

Hi,
  I have installed TE212P.  Loaded the zaptel modules  wc2xxp module for 
TE212P.
 
 The span are up  I can make a call, but the echo issue exists, so its same 
like my old TE110P card.
 
So I called Digium support.  They said that the card may be bad or the modules 
are not loaded for Hardware echo cancellor.  He said one should see Octasia  
VPM successfull message for the hardware echo cancellor to be working.
 
I get this is dmesg(which means hardware echo cancellor module is not loaded.
 
VPM400: Not Present
VPM450: Not Present
 
But I dont see any, I just see the below in dmesg.
 
TE2XXP: Setting up global serial parameters
Found a Wildcard: Wildcard TE210P (3rd Gen)
eth0: no IPv6 routers present
About to enter spanconfig!
Done with spanconfig!
Registered tone zone 0 (United States / North America)
About to enter startup!
TE2XXP: Span 1 configured for ESF/B8ZS
wct2xxp: Setting yellow alarm on span 1
SPAN 1: Primary Sync Source
VPM400: Not Present
VPM450: Not Present
Completed startup!
wct2xxp: Clearing yellow alarm on span 1
Zaptel Transcoder support loaded
 
 
Has any one had this issue with RHEL4-Update 4. Please let me know your views.
 
--
Deepak


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Re: [asterisk-users] Working softphone for poket PC

2007-05-22 Thread Arpit Mehta


Hi,

Is there any open source pocket pc softphones available. I could find only
one MiniSip that too, it was releasing soon.

Regards

Arpit

On 5/22/07, Remco Post [EMAIL PROTECTED] wrote:


Cosmin Prund wrote:
 Googling arround I found a number of pocket pc softphones. Of those I
 was only able to install SJ-something (removed it).


SJphone, and why did you remove it?

 Is there one (pocket pc softphone) that works?


SJphone ;-) At least I've made some successful calls using sjphone

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--

Remco Post

I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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--
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University

Tel: 1-646-387-5998
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[asterisk-users] FXS + Pots Extensions Help

2007-05-22 Thread Rob Schall

Hello all,

Normally I just use pri's with our asterisk systems, but a request came
in to add some normal pots lines to the setup. We have 3 lines, and they
run into the fxs ports. They hit the dialplan just fine, and they always
dial the s extension. However, my question would be... Is there a way
to determine what number was dialed and have it forward to a specific
phone? With a pri, it automatically grabs the last 4 digits, so you
could dial a number xxx-xxx-5053 and it would dial to the 5053 extension.

Any thoughts, or is this even possible?

Rob
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RE: [asterisk-users] Zaptel hangs machine...

2007-05-22 Thread Jim Suber

I'm fairly certain that zaptel is not a service.
You might try service asterisk stop
I 'm running CentOS 4.4 as well. The only problem I'm having is sometimes
the extensions don't ring even though the caller hears a ring tone.
I'm thinking maybe it's the fact that I got 5 people on one little
DSL connection around here. Downloading, playing music etc.
If you have any suggestions for me let me know.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angel Luis
Martinez
Sent: Tuesday, May 22, 2007 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zaptel hangs machine...

Tzafrir Cohen escribió:
 On Sat, May 19, 2007 at 07:26:33PM +0200, Angel Luis Martinez wrote:
   
 Hi all.

 When i do a service zaptel stop on my machine,sometimes it crash and i
 must unplug and plug the power cord to restart the machine. Also
sometimes
 load zttranscode and wct4xxp, and oter times wct4xxp only... it's running
 centos 4.4, libpri 1.2.4, zaptel 1.2.17.1 and asterisk 1.2.18. with a
 TE212P.

 Can somebody help me?
 

 Try editing /etc/init.d/zaptel and remove 'ztcfg -s' from there.

   
Thanks for you response. It works !!!

Thanks a lot !!
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[asterisk-users] (no subject)

2007-05-22 Thread Gommidh Riadh

Hello,

Did someone have a solution for a line fax detection for outgoing call

For exemple

I call number 0123456789
- if it is a fax then redirect to extension A
- if it is a line then redirect to exention B

whats ia want its somthing like AMD application that i use for the
answering machine .

http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD

search in the wiki give  this application :

http://www.voip-info.org/wiki/view/NVFaxDetect

Did somene use it ? any feed back ?

Sorry for the English and thanks for your help

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[asterisk-users] Fax detection

2007-05-22 Thread Gommidh Riadh

Hello,

Did someone have a solution for a line fax detection for outgoing call

For exemple

I call number 0123456789
- if it is a fax then redirect to extension A
- if it is a line then redirect to exention B

whats ia want its somthing like AMD application that i use for the
answering machine .

http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD

search in the wiki give  this application :

http://www.voip-info.org/wiki/view/NVFaxDetect

Did somene use it ? any feed back ?

Sorry for the English and thanks for your help

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Re: [asterisk-users] Zaptel hangs machine...

2007-05-22 Thread Anthony Francis


Jim Suber wrote:

I'm fairly certain that zaptel is not a service.
You might try service asterisk stop
I 'm running CentOS 4.4 as well. The only problem I'm having is sometimes
the extensions don't ring even though the caller hears a ring tone.
I'm thinking maybe it's the fact that I got 5 people on one little
DSL connection around here. Downloading, playing music etc.
If you have any suggestions for me let me know.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angel Luis
Martinez
Sent: Tuesday, May 22, 2007 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zaptel hangs machine...

Tzafrir Cohen escribió:
  

On Sat, May 19, 2007 at 07:26:33PM +0200, Angel Luis Martinez wrote:
  


Hi all.

When i do a service zaptel stop on my machine,sometimes it crash and i
must unplug and plug the power cord to restart the machine. Also
  

sometimes
  

load zttranscode and wct4xxp, and oter times wct4xxp only... it's running
centos 4.4, libpri 1.2.4, zaptel 1.2.17.1 and asterisk 1.2.18. with a
TE212P.

Can somebody help me?

  

Try editing /etc/init.d/zaptel and remove 'ztcfg -s' from there.

  


Thanks for you response. It works !!!

Thanks a lot !!
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zaptel is a service if you added it as one, and you should have.
To make it be a service in centos from the src dir of zaptel do:

cp zaptel.init /etc/init.d/zaptel
chmod 755 /etc/init.d/zaptel
make install-udev
echo TELEPHONY=yes  /etc/sysconfig/zaptel
TELEPHONY=yes
chkconfig --add zaptel
service zaptel start


this runs modprobe, ztcfg and all of that for you.

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Re: [asterisk-users] Delete voicemails after X days

2007-05-22 Thread Eric \ManxPower\ Wieling


David Florella wrote:

Thank you knox. Finally, I have chosen this solution : find
/var/spool/asterisk/voicemail/default/*/Old/ -atime -7|xargs rm –f, executed
every night by the CRON. However, I would have preferred this feature was
implemented in Astrisk.


You should expect this to massively break voice mailboxes.

Asterisk Voicemail requires that all messages are numbered sequentially 
starting at 0 (when using the filesystem, I don't know about RealTime or 
IMAP).  If there is a break in the sequence, such as would be the case 
if your script deletes a message in the middle, then you should expect 
things to break.  I think that higher numbered messages would simply not 
be accessible, but that is a guess.


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Re: [asterisk-users] Re: DUNDi configuration problem

2007-05-22 Thread Tim Verscheure


Does this even work?

exten = 5010,1,Dial(SIP/[EMAIL PROTECTED])

It keeps saying CHANUNAVAIL...


greetz

2007/5/22, Tim Verscheure [EMAIL PROTECTED]:

ok so now I changed ext-local to dundi-ext and I created this
context at the bottom of the extensions file. This is now the case.

[dundi-priv-canonical]
; Direct numbers

exten = 5010,1,NooP(DUNDI LOOKUP 5010)
exten = 5011,1,NooP(DUNDI LOOKUP 5011)

exten = _60XX,1,Goto(dundi-ext,${EXTEN},1)

[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
exten = _60XX,1,Goto(dundi-ext,${EXTEN},1)

[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here

[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn

exten = 5010,1,Dial(SIP/5010)
exten = 5011,1,Dial(SIP/5011)

[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv

[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch

[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-lookup

[trydundi]
exten = _.,1,Macro(dundi-priv,${EXTEN})
exten = _.,2,Congestion


This is the dundi-ext at the bottom. In there I put this line:
[dundi-ext]
exten = _60XX,1,Dial(SIP/[EMAIL PROTECTED])

I made this myself, I think that if I get an incoming call from for
example 6010, the person would be dialing SIP/[EMAIL PROTECTED], right?

this is the output:
*CLI -- Executing [EMAIL PROTECTED]:1] Goto(SIP/6010-0820cdc8,
dundi-ext|5011|1) in new stack
-- Goto (dundi-ext,5011,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/6010-0820cdc8,
SIP/[EMAIL PROTECTED]) in new stack
[May 22 14:36:18] WARNING[4520]: chan_sip.c:2738 create_addr: No such host: priv
[May 22 14:36:18] WARNING[4520]: app_dial.c:1099 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/6010-0820cdc8' status is 'CHANUNAVAIL'


2007/5/21, Remco Post [EMAIL PROTECTED]:
 Tim Verscheure wrote:
  Now I get this... If I call from 5011 on the 192.168.1.103 machine to
  6010 on the 192.168.1.69 machine my X-lite softphone says, call
  declined
 
  this is the output:
 -- Executing [EMAIL PROTECTED]:1] Goto(SIP/5011-081da508,
  ext-local|6010|1) in new stack
 -- Goto (ext-local,6010,1)
  [May 21 15:32:46] WARNING[8939]: pbx.c:2450 __ast_pbx_run: Channel
  'SIP/5011-081da508' sent into invalid extension '6010' in context
  'ext-local', but no invalid handler
 

 so, is there an extension 6010 in you context ext-local? Probably not ;-)

  I'll add my extension file so you can see it. greetz


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Re: [asterisk-users] Delete voicemails after X days

2007-05-22 Thread Luki


You should expect this to massively break voice mailboxes.


Well, it won't massively break them, just a bit. We do this on some
mailboxes and it works OK. The problem is that is you delete message 1
and leave 2, a new message will become 1, thus breaking the sequence.
They will be played back as 1 (newer) followed by 2 (older) message.
Then again, I'm not sure what happens if there is a break in sequence
-- I think I patched my code to deal with that. It's ugly and
inefficient.

Still all of these solutions are a band aid at best. I don't like do
it this way. I wish Asterisk could do it itself.

--Luki
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[asterisk-users] Mix Dial, Chanspy and MixMonitor or Monitor

2007-05-22 Thread ennead-70866

I have an application that requires I be able to dial into an asterisk box, 
then from there dial out to another user through a PSTN.  I'd like to be able 
to both 1) record this call and 2) let another user dial in using something 
like ChanSpy to listen to the conversation.

I can get this working by executing an auto-dial script to connect one end of a 
call to an outside Asterisk box which does the recording, and the local end 
which listens in via ChanSpy.  Another user could then dial in and also listen 
via ChanSpy.

The problem with this is that it's very clunky, and I'd like to keep everything 
local.  Problem is that when I try to use Dial, Chanspy and MixMonitor I get no 
audio, which is why I do it on the outside Asterisk box.

Here's a basic framework:

;;Main Asterisk Box
[inbound]
exten = dialout,1,Set(SPYGROUP=10001)
exten = dialout,2,set(ALLREAD=555777)
exten = dialout,3,dial(SIP/[EMAIL PROTECTED])
exten = dialout,4,hangup

[listen-in]
; inbound portion of autodial or
; outside caller
exten = monitor,1,answer
exten = monitor,2,chanspy(all|qg(10001))
exten = monitor,3,hangup

;;Outside Asterisk Box
[auto-dial-remote]
 ; call initiated by autodial
exten = s,1,answer
exten = s,2,mixmonitor(/tmp/test.wav)
exten = s,3,hangup

Any help would be appreciated.

Best regards,

Klive
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Re: [asterisk-users] Delete voicemails after X days

2007-05-22 Thread Joel Hill

Sorry to say I have to disagree with you but I just had a heap of old
Voicemails which I couldn't be bothered deleting through my phone, So I
went in to /Old/ and ran rm -f on the first 20, I then had to listen to
another that wasn't deleted and it was still accessible from the phone,
upon further investigation asterisk has renamed them starting again at
0. So running a CRON job to do the same thing should work fine.

Cheers,

Joel

On Tue, 2007-05-22 at 20:37 -0500, Eric ManxPower Wieling wrote:
 David Florella wrote:
  Thank you knox. Finally, I have chosen this solution : find
  /var/spool/asterisk/voicemail/default/*/Old/ -atime -7|xargs rm –f, executed
  every night by the CRON. However, I would have preferred this feature was
  implemented in Astrisk.
 
 You should expect this to massively break voice mailboxes.
 
 Asterisk Voicemail requires that all messages are numbered sequentially 
 starting at 0 (when using the filesystem, I don't know about RealTime or 
 IMAP).  If there is a break in the sequence, such as would be the case 
 if your script deletes a message in the middle, then you should expect 
 things to break.  I think that higher numbered messages would simply not 
 be accessible, but that is a guess.
 
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Re: [asterisk-users] Zaptel hangs machine...


Anthony Francis wrote:

Jim Suber wrote:

I'm fairly certain that zaptel is not a service.
You might try service asterisk stop
I 'm running CentOS 4.4 as well. The only problem I'm having is sometimes
the extensions don't ring even though the caller hears a ring tone.
I'm thinking maybe it's the fact that I got 5 people on one little
DSL connection around here. Downloading, playing music etc.
If you have any suggestions for me let me know.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angel Luis
Martinez
Sent: Tuesday, May 22, 2007 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zaptel hangs machine...

Tzafrir Cohen escribió:
 

On Sat, May 19, 2007 at 07:26:33PM +0200, Angel Luis Martinez wrote:
 

Hi all.

When i do a service zaptel stop on my machine,sometimes it crash 
and i

must unplug and plug the power cord to restart the machine. Also
  

sometimes
 
load zttranscode and wct4xxp, and oter times wct4xxp only... it's 
running

centos 4.4, libpri 1.2.4, zaptel 1.2.17.1 and asterisk 1.2.18. with a
TE212P.

Can somebody help me?
  

Try editing /etc/init.d/zaptel and remove 'ztcfg -s' from there.

  

Thanks for you response. It works !!!

Thanks a lot !!
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zaptel is a service if you added it as one, and you should have.
To make it be a service in centos from the src dir of zaptel do:

cp zaptel.init /etc/init.d/zaptel
chmod 755 /etc/init.d/zaptel
make install-udev
echo TELEPHONY=yes  /etc/sysconfig/zaptel
TELEPHONY=yes
chkconfig --add zaptel
service zaptel start



Nice little tidbit.  I've added this to my notes for future reference.

Thanks for sharing.

--

Warm Regards,

Lee



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Re: [asterisk-users] Phones fail to ring