[Asterisk-Users] TE405P Dropping Calls

[James Bean]

Hi,

Urgently response would be wonderful, system is a Fedora Core 2.

I have a Ericsson BP250 connected to 1 port on the TE405P and another
connected to a local telco ISDN30.

I have been running CVS-HEAD from about a 2 months ago and upgraded it
again just in cause it was a version issue (didn't fix it) but this is
what I am getting.

When a person calls out from an extension on the BP250 to the outside
world, every 2-5 minutes randomly it drops all active calls with the
following error on console.

 == Spawn extension (te405p-intelstra, 38166483, 2) exited non-zero on
'Zap/5-1'
-- Hungup 'Zap/2-1'
  == Spawn extension (te405p-frombp250, 00402270883, 1) exited non-zero
on 'Zap/95-1'
-- Hungup 'Zap/98-1'
  == Spawn extension (te405p-intelstra, 38166483, 2) exited non-zero on
'Zap/7-1'
-- Hungup 'Zap/1-1'
  == Spawn extension (te405p-frombp250, 00417513573, 1) exited non-zero
on 'Zap/94-1'
-- Hungup 'Zap/97-1'
  == Spawn extension (te405p-intelstra, 38165965, 2) exited non-zero on
'Zap/6-1'
-- Hungup 'Zap/5-1'
-- Hungup 'Zap/95-1'
-- Hungup 'Zap/7-1'
-- Hungup 'Zap/94-1'
-- Hungup 'Zap/6-1'
!! Got reject for frame 77, retransmitting frame 77 now, updating n_r!
!! Got reject for frame 77, retransmitting frame 78 now, updating n_r!
!! Got reject for frame 77, retransmitting frame 79 now, updating n_r!
!! Got reject for frame 77, retransmitting frame 80 now, updating n_r!
!! Got reject for frame 77, retransmitting frame 81 now, updating n_r!

If I unplug the ISDN30 from asterisk and plug it directly into the BP250
it works fine no problems.

It only just started happening and there have been no changes to the
system configuration or setup for over 2 weeks, I am tempted to try and
downgrade to 1.0.7. Although the telco did add additional DID's for the
ISDN 30, which were added to the extensions.conf, but with them removed
it still did the same thing.

My configuration files are as follows :-

/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
span=3,1,0,ccs,hdb3,crc4
bchan=63-77
dchan=78
bchan=79-93
span=4,1,0,ccs,hdb3,crc4
bchan=94-108
dchan=109
bchan=110-124

loadzone=au

/etc/asterisk/zapata.conf
[channels]
context=default
musiconhold=default
switchtype=euroisdn
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0.0
txgain=0.0

group=1
context=te405p-intelstra
;context=te405p-ext
pridialplan=local
signalling=pri_cpe
;overlapdial=yes
callerid=asreceived
channel=1-15, 17-31

group=4
context=te405p-frombp250
;context=te405p-in
pridialplan=local
signalling=pri_net
overlapdial=yes
callerid=asreceived
channel=94-108, 110-124

/etc/asterisk/extensions.conf
[dialstring]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

[te405p-frombp250]

include = to-sip
include = te405p-outtelstra

[te405p-tobp250]

exten = _2XX,1,Dial(Zap/g4/${EXTEN},60,r)
exten = _4XX,1,Dial(Zap/g4/${EXTEN},60,r)
exten = _7XX,1,Dial(Zap/g4/${EXTEN},60,r)
exten = _1XXX,1,Dial(Zap/g4/${EXTEN},60,r)
exten = _73816592XX,1,Dial(Zap/g4/${EXTEN:7},60,r)

[te405p-intelstra]

exten = s,1,SetMusicOnHold(record)
exten = s,2,Dial(SIP/bt-pavilion,45,t)
exten = s,4,VoiceMail,u500
exten = s,5,Hangup

exten = 38166400,1,SetMusicOnHold(random)
exten = 38166400,2,Dial(Zap/g4/9,600,t)
exten = 38166400,3,VoiceMail,u500
exten = 38166400,4,Hangup

;Main Number 3282 2922 comes in on 38166483
exten = 38166483,1,SetMusicOnHold(random)
exten = 38166483,2,Dial(Zap/g4/9,6000,t)
exten = 38166483,3,VoiceMail,u500
exten = 38166483,4,Hangup

;Tempory Number for Darryl
exten = 38166488,1,SetMusicOnHold(random)
exten = 38166488,2,Dial(Zap/g4/483,6000,t)
exten = 38166488,3,VoiceMail,u500
exten = 38166488,4,Hangup

exten = 32822922,1,SetMusicOnHold(random)
exten = 32822922,2,Dial(Zap/g4/9,600,t)
exten = 32822922,3,VoiceMail,u500
exten = 32822922,4,Hangup

;exten =
38166444,1,SetVar(CALLFILENAME=/mnt/asterisk/38166444/${CALLERID}-${TIME
STAMP})
;exten = 38166444,2,Monitor(gsm,${CALLFILENAME},m)
exten = 38166444,1,Dial(Zap/g4/9)

;exten =
38166483,1,SetVar(CALLFILENAME=/mnt/asterisk/38166483/${CALLERID}-${TIME
STAMP})
;exten = 38166483,2,Monitor(gsm,${CALLFILENAME},m)
;exten = 38166483,3,Dial(Zap/g4/483)

exten = _381684XX,1,SetMusicOnHold(record)
exten = _381684XX,2,Dial(Zap/g4/2${EXTEN:-2},600,t)
exten = _381684XX,3,VoiceMail,u500
exten = _381684XX,4,Hangup

exten = _381659XX,1,SetMusicOnHold(record)
exten = _381659XX,2,Dial(Zap/g4/2${EXTEN:-2},6000,t)
exten = _381659XX,3,Dial(Zap/g4/211,600,t)
exten = _381659XX,4,Hangup

[te405p-outtelstra]

exten = _00011X.,1,AGI(blockintl.agi|${EXTEN:1})

exten = _01902X.,1,Hangup

exten = _0X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _0X.,2,Congestion
exten = _0X.,3,Hangup

include = dialstring

[to-sip]

RE: [Asterisk-Users] TE405P Dropping Calls

[James Bean]

Update...

Figured out it was a faulty port in the te405p, swapped to a spare port
and all it good, now to get warranty.

James 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Bean
Sent: Saturday, 6 August 2005 10:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] TE405P Dropping Calls


Hi,

Urgently response would be wonderful, system is a Fedora Core 2.

I have a Ericsson BP250 connected to 1 port on the TE405P and another
connected to a local telco ISDN30.

I have been running CVS-HEAD from about a 2 months ago and upgraded it
again just in cause it was a version issue (didn't fix it) but this is
what I am getting.

When a person calls out from an extension on the BP250 to the outside
world, every 2-5 minutes randomly it drops all active calls with the
following error on console.

 == Spawn extension (te405p-intelstra, 38166483, 2) exited non-zero on
'Zap/5-1'
-- Hungup 'Zap/2-1'
  == Spawn extension (te405p-frombp250, 00402270883, 1) exited non-zero
on 'Zap/95-1'
-- Hungup 'Zap/98-1'
  == Spawn extension (te405p-intelstra, 38166483, 2) exited non-zero on
'Zap/7-1'
-- Hungup 'Zap/1-1'
  == Spawn extension (te405p-frombp250, 00417513573, 1) exited non-zero
on 'Zap/94-1'
-- Hungup 'Zap/97-1'
  == Spawn extension (te405p-intelstra, 38165965, 2) exited non-zero on
'Zap/6-1'
-- Hungup 'Zap/5-1'
-- Hungup 'Zap/95-1'
-- Hungup 'Zap/7-1'
-- Hungup 'Zap/94-1'
-- Hungup 'Zap/6-1'
!! Got reject for frame 77, retransmitting frame 77 now, updating n_r!
!! Got reject for frame 77, retransmitting frame 78 now, updating n_r!
!! Got reject for frame 77, retransmitting frame 79 now, updating n_r!
!! Got reject for frame 77, retransmitting frame 80 now, updating n_r!
!! Got reject for frame 77, retransmitting frame 81 now, updating n_r!

If I unplug the ISDN30 from asterisk and plug it directly into the BP250
it works fine no problems.

It only just started happening and there have been no changes to the
system configuration or setup for over 2 weeks, I am tempted to try and
downgrade to 1.0.7. Although the telco did add additional DID's for the
ISDN 30, which were added to the extensions.conf, but with them removed
it still did the same thing.

My configuration files are as follows :-

/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
span=3,1,0,ccs,hdb3,crc4
bchan=63-77
dchan=78
bchan=79-93
span=4,1,0,ccs,hdb3,crc4
bchan=94-108
dchan=109
bchan=110-124

loadzone=au

/etc/asterisk/zapata.conf
[channels]
context=default
musiconhold=default
switchtype=euroisdn
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0.0
txgain=0.0

group=1
context=te405p-intelstra
;context=te405p-ext
pridialplan=local
signalling=pri_cpe
;overlapdial=yes
callerid=asreceived
channel=1-15, 17-31

group=4
context=te405p-frombp250
;context=te405p-in
pridialplan=local
signalling=pri_net
overlapdial=yes
callerid=asreceived
channel=94-108, 110-124

/etc/asterisk/extensions.conf
[dialstring]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

[te405p-frombp250]

include = to-sip
include = te405p-outtelstra

[te405p-tobp250]

exten = _2XX,1,Dial(Zap/g4/${EXTEN},60,r) exten =
_4XX,1,Dial(Zap/g4/${EXTEN},60,r) exten =
_7XX,1,Dial(Zap/g4/${EXTEN},60,r) exten =
_1XXX,1,Dial(Zap/g4/${EXTEN},60,r)
exten = _73816592XX,1,Dial(Zap/g4/${EXTEN:7},60,r)

[te405p-intelstra]

exten = s,1,SetMusicOnHold(record)
exten = s,2,Dial(SIP/bt-pavilion,45,t)
exten = s,4,VoiceMail,u500
exten = s,5,Hangup

exten = 38166400,1,SetMusicOnHold(random) exten =
38166400,2,Dial(Zap/g4/9,600,t) exten = 38166400,3,VoiceMail,u500 exten
= 38166400,4,Hangup

;Main Number 3282 2922 comes in on 38166483 exten =
38166483,1,SetMusicOnHold(random) exten =
38166483,2,Dial(Zap/g4/9,6000,t) exten = 38166483,3,VoiceMail,u500
exten = 38166483,4,Hangup

;Tempory Number for Darryl
exten = 38166488,1,SetMusicOnHold(random) exten =
38166488,2,Dial(Zap/g4/483,6000,t)
exten = 38166488,3,VoiceMail,u500
exten = 38166488,4,Hangup

exten = 32822922,1,SetMusicOnHold(random) exten =
32822922,2,Dial(Zap/g4/9,600,t) exten = 32822922,3,VoiceMail,u500 exten
= 32822922,4,Hangup

;exten =
38166444,1,SetVar(CALLFILENAME=/mnt/asterisk/38166444/${CALLERID}-${TIME
STAMP})
;exten = 38166444,2,Monitor(gsm,${CALLFILENAME},m)
exten = 38166444,1,Dial(Zap/g4/9)

;exten =
38166483,1,SetVar(CALLFILENAME=/mnt/asterisk/38166483/${CALLERID}-${TIME
STAMP})
;exten = 38166483,2,Monitor(gsm,${CALLFILENAME},m)
;exten = 38166483,3,Dial(Zap/g4/483)

exten = _381684XX,1,SetMusicOnHold(record)
exten = _381684XX,2,Dial(Zap/g4/2${EXTEN:-2},600,t)
exten = _381684XX,3,VoiceMail,u500
exten = _381684XX,4,Hangup

exten

[Asterisk-Users] Asterisk slow transferring calls

[James Bean]

Hi, 

Running Asterisk CVS-Head latest on a Dual P3 800 1Gb Ram.

For some odd reason now that I have the asterisk box almost to the stage
I want it, I hit a problem.

I have a te405p in the system, Zap/g1 is connected to the telco as an
ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an Ericcson BP250
phone system.

When calls come in on g1 they go straight through instantaneously to the
entensions on the ericsson or local sip phones no problems, if someone
on a sip phone calls an extension on the ericsson it goes straight
through no pause.

If someone on the ericsson system dials a sip phone it takes close to 3
full seconds before the sip phone rings, it takes that long just to get
to the asterisk box, although its not the ericsson phone system that is
the problem, if I dump a straight plain extensions.conf into the system
it works perfectly and is fast from the ericsson to the sip phone, if I
use the one I want to get running its slow again.

Can someone have a breeze through and let me know what they think might
be causing the problem.

I think I am not getting the right idea with out the contexts work and
it might be looping or something, te405p-in and sip need access to each
other and the ability to dialout, and voip, voip needs access to dial
the ericsson system and the sip phones (haven't added that part yet) but
not access to an outside line.

James

My extensions.conf

#include extensions_sip.conf

[globals]
EMERGENCY=0
EMERGENCY_TRUNK=Zap/10

[dialstring]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

[atp-out]

exten =
_9X.,1,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN:1})
exten = _9X.,2,Congestion
exten = _9X.,3,Hangup

[atp-in]

exten = 30182849,1,SetMusicOnHold(record)
exten = 30182849,2,Dial(SIP/bt-rlm,45,t)
exten = 30182849,3,Voicemail,u550
exten = 30182849,103,Voicemail,b550

[te405p-in]

exten = _2XX,1,Dial(Zap/g4/${EXTEN},60,r)
exten = _2XX,2,Hangup

exten = _73816592XX,1,Dial(Zap/g4/${EXTEN:-3},60,r)
exten = _73816592XX,2,Hangup

exten = _7XX,1,Dial(Zap/g4/${EXTEN},60,r)
exten = _7XX,2,Hangup

exten = _1XXX,1,Dial(Zap/g4/${EXTEN},60,r)
exten = _1XXX,2,Hangup

include = sip
include = parkedcalls
include = te405p-outgoing
include = transfer-record

[te405p-ext]

exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/bt-pavilion,45,t)
exten = s,4,VoiceMail,u500
exten = s,5,Hangup

exten = 38166400,1,SetMusicOnHold(random)
exten = 38166400,2,Dial(Zap/g4/211,600,t)
exten = 38166400,3,VoiceMail,u500
exten = 38166400,4,Hangup

exten = 38166444,1,DISA(1234|sip)

exten = _381664XX,1,SetMusicOnHold(random)
exten = _381664XX,2,Dial(Zap/g4/2${EXTEN:-2},600,t)
exten = _381664XX,3,VoiceMail,u500
exten = _381664XX,4,Hangup

[te405p-outgoing]

exten =
000,1,SetVar(CALLFILENAME=/mnt/asterisk/EMERGENCY_CALL-${CALLERID}-${TIM
ESTAMP})
exten = 000,2,Monitor(gsm,${CALLFILENAME},m)
exten = 000,3,Goto(emergency,s,1)

exten =
,1,SetVar(CALLFILENAME=/mnt/asterisk/EMERGENCY_CALL-${CALLERID}-${TI
MESTAMP})
exten = ,2,Monitor(gsm,${CALLFILENAME},m)
exten = ,3,Goto(emergency,s,1)

exten = _00011X.,1,AGI(blockintl.agi|${EXTEN:1})

exten = _01902X.,1,Hangup

exten =
_0X.,1,SetVar(CALLFILENAME=/mnt/asterisk/${CALLERID}-${EXTEN:1}-${TIMEST
AMP})
exten = _0X.,2,Monitor(gsm,${CALLFILENAME},m)
exten = _0X.,3,Dial(Zap/g1/${EXTEN:1})
exten = _0X.,4,Congestion
exten = _0X.,5,Hangup

include = phatphingers

[transfer-record]

exten =
_52XX,1,SetVar(CALLFILENAME=/mnt/asterisk/CallTo-${EXTEN:1}-${TIMESTAMP}
)
exten = _52XX,2,Monitor(gsm,${CALLFILENAME},m)
exten = _52XX,3,Dial(ZAP/g4/${EXTEN:1})
exten = _52XX,4,Congestion
exten = _52XX,104,Congestion

[voip]

exten = 589,1,Dial(IAX2/username:[EMAIL PROTECTED]/690)
exten = _2XX,1,Dial(Zap/g4/${EXTEN},60,r)

[parkedcalls]

exten = 590,1,playback(lm1/call_may_be_recorded)
exten =
590,2,ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/DNE|te405p-in,Zap/g4/
211,1)

[emergency]
exten = s,1,SetVar(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,SetGlobalVar(EMERGENCY=1)
exten = s,n,SetVar(SET_EMERG_FLAG=1)
exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)
exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
exten = s,n,Wait(12)
exten = s,n,Goto(checkavail)
exten = s,s+2(inprogress),Congestion
exten = s,checkavail+101(notavail),Goto(trunkbusy)
exten = h,1,GotoIf($[${SET_EMERG_FLAG} = 1]?3)
exten = h,3,SetGlobalVar(EMERGENCY=0)

[phatphingers]
exten = _X.,1,answer
exten = _X.,2,wait(.5)
exten = _X.,3,playback(vm-extension)
exten = _X.,4,sayalpha(${EXTEN})
exten = _X.,5,playback(invalid)
exten = _X.,6,hangup

My extensions_sip.conf

[sip]

exten = 555,1,SetMusicOnHold(random)
exten = 555,2,Dial(ZAP/g4/211)
exten = 555,3,Voicemail,u555
exten = 555,103,Voicemail,b555

exten = 556,1,SetMusicOnHold(random)
exten = 556,2,Dial(SIP/js-softphone,30,Ttr)
exten = 556,3,Voicemail,u556
exten = 556,103,Voicemail,b556

exten = 557,1,SetMusicOnHold(random)

[Asterisk-Users] Help with denighing access to certain numbers by CallerID

[James Bean]

Hi,

Asterisk 1.0.7
TE405P - Port 1 - ISDN30 telco
   - Port 4 - Primary Rate connection to Phone system

The system has a mixture of 20+ sip phones and the 50 odd extensions on
the phone system connected to Port 4.

What I want to accomplish is to be able to denigh access to certain
outgoing phone calls by the extension/callerid the call originated from.

i.e. Only certain sip and telephone extensions from the phone system can
dial 0011 (international) numbers, and well I already have the porn/pay
per call 1902 calls blocked already.

The difficulty I am having is a problem with being able to block by the
callerid from the phone system calls, I get the callerid from the
incoming extension no problems but blocking certain numbers by it is the
problem.

I can only really see 2 ways to do it, and it's a little above me and I
was hoping someone has done it and I can get a copy of what they did :-
1. Some sort of api or script in the checks the callerid against an
authorised list.
2. setup up the restricted numbers can only be used with a secret code,
when the number is dialed it prompts the users to punch in an access
code to allow the call.

Any suggestions, anyone already done it can throw me an example,
hopefully trying to avoid several pages of allows for the people who can
uses those numbers.

James
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RE: [Asterisk-Users] Help with denighing access to certain numbersbyCallerID

[James Bean]

Sounds like what I had in mind, could you point me in the right
direction to an example of agi scripts that might do this :-). I'm not
well versed in the ways of the AGI and would flounder significantly.

James

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Thomas
Sent: Sunday, 12 June 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with denighing access to certain
numbersbyCallerID

I would think the easiest way is to use an AGI..

exten = _001162.,Dial(Zap/g1/${EXTEN}/tT); (Anyone can call NZ) exten
= _0011.,Exec(checkperms);

checkperms.agi would then match against a list.

--Rob


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users- 
 [EMAIL PROTECTED] On Behalf Of James Bean
 Sent: Sunday, June 12, 2005 8:40 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Help with denighing access to certain
numbers
 byCallerID
 
 
 Hi,
 
 Asterisk 1.0.7
 TE405P - Port 1 - ISDN30 telco
- Port 4 - Primary Rate connection to Phone system
 
 The system has a mixture of 20+ sip phones and the 50 odd extensions
on
 the phone system connected to Port 4.
 
 What I want to accomplish is to be able to denigh access to certain 
 outgoing phone calls by the extension/callerid the call originated
from.
 
 i.e. Only certain sip and telephone extensions from the phone system
can
 dial 0011 (international) numbers, and well I already have the
porn/pay
 per call 1902 calls blocked already.
 
 The difficulty I am having is a problem with being able to block by
the
 callerid from the phone system calls, I get the callerid from the 
 incoming extension no problems but blocking certain numbers by it is
the
 problem.
 
 I can only really see 2 ways to do it, and it's a little above me and
I
 was hoping someone has done it and I can get a copy of what they did
:-
 1. Some sort of api or script in the checks the callerid against an 
 authorised list.
 2. setup up the restricted numbers can only be used with a secret
code,
 when the number is dialed it prompts the users to punch in an access 
 code to allow the call.
 
 Any suggestions, anyone already done it can throw me an example, 
 hopefully trying to avoid several pages of allows for the people who
can
 uses those numbers.
 
 James
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RE: [Asterisk-Users] GXP2000 and hint LED's

[James Bean]

Bugger, thanks for replying and telling me, might send a request through
to Grandstream and see when they intend on releasing it.

James 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Svensson
Sent: Thursday, 9 June 2005 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP2000 and hint LED's

On Thu, 9 Jun 2005, James Bean wrote:

 Has anyone got the hint function working, and maybe with the GXP2000.

I don't think the current firmware release for the GXP-2000 supports
SUBSCRIBE/NOTIFY. That functionality is to be released at a later date.

Peter


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RE: [Asterisk-Users] GXP2000 and hint LED's

[James Bean]

Did that pre-release version fix that bug where the other party can
hear you when you pressed the transfer button ?
Does it also enable the leds next to the speeddial buttons like the
snoms ?


Unfortunately not, Grandstream didn't admit to me that they were going
to program the LED's like the snom SUBCRIBE/NOTIFY, they told me the
LED's were additional incoming line indicators, not LED's for the
function keys to be programmed. Which is a little stupid, if they don't
do the LED's like the snom then the phone is really no better then the
BT102, just with a bigger LED and multiple sip account capability.

If you want the 1.0.1.9 firmware pre-release goto www.atp.org.au and on
the main page near the bottom it gives you a link.

Peter seems to be on the ball more then me about these phones as
grandstream gave me the standard replies, Peter do you know for sure if
grandstream have a timetable for the function led's cause I need to
rollout about 50 phones and need 6-7 led's for display, which means a
snom220+expansion, and gxp2000 seems perfect if it worked.

James
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[Asterisk-Users] Parked Call queue function key notify

[James Bean]

Does anyone know if the parked call queue has hint built into it.

I want to program up on a series of sip phones that support subscribe
notify on the led function keys the parked call queue (791-795)
positions so that its easy for people to know there is someone in the
queue and the calls can be picked up easily.

Reception just puts them in the parked queue, does a broadcast call to
say Blah is on PQ1 or PQ2 and the person can just hit the button to pick
them up.

James



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RE: [Asterisk-Users] GXP2000 and hint LED's

[James Bean]

Unfortunately the cost is excessive on the snom360 in Australia about
$470 where the GXP2000 is $180, that's a huge difference when dealing
with 50+ phones.

Just have to wait the 6-8 weeks for grandstreams first attempt at
getting it working.

James

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of The VoIP
Connection
Sent: Friday, 10 June 2005 9:46 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] GXP2000 and hint LED's

James,

If you don't think you want to wait for the Grandstream, the snom 360
will do what you need (12 programmable buttons).  We are offering great
pricing on these right now.

http://www.thevoipconnection.com/store/catalog/product_16234_snom_360_Ex
ecut
ive_IP_Telephone.html

Michael Crown
Managing Partner
The VoIP Connection
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
 

 -Original Message-
 From: James Bean [mailto:[EMAIL PROTECTED]
 Sent: Thursday, June 09, 2005 5:03 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] GXP2000 and hint LED's
 
 
 Did that pre-release version fix that bug where the other party can
 hear you when you pressed the transfer button ?
 Does it also enable the leds next to the speeddial buttons like the
 snoms ?
 
 
 Unfortunately not, Grandstream didn't admit to me that they were going

 to program the LED's like the snom SUBCRIBE/NOTIFY, they told me the 
 LED's were additional incoming line indicators, not LED's for the 
 function keys to be programmed. Which is a little stupid, if they 
 don't do the LED's like the snom then the phone is really no better 
 then the BT102, just with a bigger LED and multiple sip account 
 capability.
 
 If you want the 1.0.1.9 firmware pre-release goto www.atp.org.au and 
 on the main page near the bottom it gives you a link.
 
 Peter seems to be on the ball more then me about these phones as 
 grandstream gave me the standard replies, Peter do you know for sure 
 if grandstream have a timetable for the function led's cause I need to

 rollout about 50 phones and need 6-7 led's for display, which means a
 snom220+expansion, and gxp2000 seems perfect if it worked.
 
 James
 
 

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RE: [Asterisk-Users] Clicks in audio with TE100P PRI

[James Bean]

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
G
Sent: Thursday, 9 June 2005 1:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Clicks in audio with TE100P PRI


Thanks for your answer. Googling in the lists I found what you are
telling that maybe there is a synchro problem with the E1, but I'm not
so sure that this could be. I am configuring zaptel.conf like this:

span=1,0,0,ccs,hdb3
bchan=1-15,17-31
dchan=16

But I also changed to test to:

span=1,1,0,ccs,hdb3

The same thing happens.

You may consider also that if I connect PAP2 to LAN everything works,
also if I use other ip phone from internet works fine.

I also check if I'm loosing interrupts and everything seems ok. Also I
pull out the TDM400 from the box.
At last I change jitterbuffer=16 and it works better, the clicks are
reduced. Could this be possible? What is the function of this parameter
in zapata.conf?

I should tell you that the TE100P is connected to another E1 board (not
a live E1) from Natural Microsystems which acts as a gateway to PSTN.
This board works as a PRI master but I don't think that this could be
the problem as long as using other phones or in LAN it works perfectly
and the voice is clear with no clicks o sound looses.


Thanks again,


Alejandro



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[Asterisk-Users] GXP2000 and hint LED's

[James Bean]

Asterisk 1.0.7

Has anyone got the hint function working, and maybe with the GXP2000.

I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment
trying to get the LED's to light up.

On ext 690, button 1 is setup for ext 691, I did this using both methods
691, and sip:[EMAIL PROTECTED]
On ext 691, button 1 is setup for ext 690, I did this using both methods
690, and sip:[EMAIL PROTECTED]

The buttons work calling each other (both methods), but when you make a
call out the light doesn't show up on the other phone.

Any suggestions on what I might have wrong.

sip.conf

[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = all
allow = ilbc
allow = alaw
allow = ulaw
nat=disable
srvlookup=no
localnet=192.168.69.0/255.255.255.0
subscribecontext = sip
incominglimit = 1

[690]
type=friend
secret=secret
host=dynamic
callerid=James Bean 690
defaultip=192.168.69.250
dtmfmode=info
mailbox=690

[691]
type=friend
secret=secret
host=dynamic
callerid=Soft Test Phone 691
defaultip=192.168.69.69
dtmfmode=info
mailbox=691

Extensions.conf

[pstn]
exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/690Zap/g2,45,t)
exten = s,4,VoiceMail,u690
exten = s,5,Hangup

[internal]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 692,1,SetMusicOnHold(random)
exten = 692,2,Dial(Zap/g2,30,Ttr)
exten = 692,3,Voicemail,u690
exten = 692,103,Voicemail,b690

exten = 098,1,WaitMusicOnHold(45)
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone

exten = 699,1,VoiceMailMain(s${CALLERIDNUM})

include = sip
include = parkedcalls
include = outgoing

[outgoing]

exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup

exten = 000,1,Dial(Zap/g1/${EXTEN:1})
exten = 000,2,Congestion()
exten = 000,3,Hangup

[sip]

exten = 690,hint,SIP/690
exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/690,30,Ttr)
exten = 690,3,Voicemail2,u690
exten = 690,103,Voicemail2,b690

exten = 691,hint,SIP/691
exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/691,30,Ttr)
exten = 691,3,Voicemail2,u690
exten = 691,103,Voicemail2,b690

include = internal
include = outgoing
include = parkedcalls
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[Asterisk-Users] IAX calls between asterisk boxes works 1 way only

[James Bean]

Hi,

Weird issue, 2 asterisk boxes running 1.0.7, when I call iax from box 1
to box 2 it works fine, when I dial from box 2 to box 1 I get a 

On Box 1
Apr 11 17:25:39 WARNING[8969]: chan_iax2.c:5553 socket_read: Call
rejected by 192.168.69.1: No authority found

On Box 2
Apr 11 17:26:07 NOTICE[2157]: chan_iax2.c:6573 socket_read: Rejected
connect attempt from 192.168.254.100, who was trying to reach '690@'

Error, so I obviously missed something and can someone smack me upside
the head and point out my error.

Please assume that the passwords are correct in the files :-).

Configurations are attached of each box:

Box 1

iax.conf

[general]
bindport=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
tos=lowdelay

[guest]
type=user
context=default
callerid=Guest IAX User

[salisbury]
type=friend
host=192.168.254.100
username=northbuild
secret=password
context=voip
permit=192.168.254.100

extensions.conf

[global]

PSTNLine=Zap/g1
AnalogPhone=Zap/g2

[pstn]

exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
exten = s,4,VoiceMail(u690)
exten = s,5,Hangup

[internal]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 098,1,WaitMusicOnHold(45)
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone

exten = 1690,1,VoicemailMain,s690
exten = 1691,1,VoicemailMain,s691

[outgoing]

exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup

[sip]

exten = 690,hint,SIP/snom-james
exten = 691,hint,SIP/bt-karen

exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/snom-james,30,Ttr)
exten = 690,3,Voicemail2,u690
exten = 690,103,Voicemail2,b690

exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,Ttr)
exten = 691,3,Voicemail,u691
exten = 691,103,Voicemail,b691

include = internal
include = outgoing
include = parkedcalls
include = voip

[voip]

exten = _1XX,1,Dial(IAX2/james:password@192.168.254.100/${EXTEN})
exten = _65X,1,Dial(IAX2/dixst:password@192.168.50.1/${EXTEN})

-
Box 2

iax.conf

[general]
bindport=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
tos=lowdelay

[guest]
type=user
context=default
callerid=Guest IAX User

[dixst]
type=friend
host=192.168.50.1
username=dixst
secret=password
context=e100p
permit=192.168.50.1

[james]
type=friend
host=192.168.69.1
username=james
secret=password
context=e100p
permit=192.168.69.1

extensions.conf

[dialstring]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

[e100p]

exten = _1XX,1,Dial(Zap/g1/${EXTEN})
exten = _93X.,1,Dial(Zap/g1/${EXTEN})
exten = _9073X.,1,Dial(Zap/g1/${EXTEN})

include = dialstring
include = voip

[voip]

exten = _65X,1,Dial(IAX2/dixst:password@192.168.50.1/${EXTEN})
; DixSt Redcliffe Ext
exten = _66X,1,Dial(IAX2/scarb:password@192.168.60.1/${EXTEN})
; Scarborough
exten = _69X,1,Dial(IAX2/northbuild:password@192.168.69.1/${EXTEN})
; James Home
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[Asterisk-Users] Supply ringing noise to IAX callers

[James Bean]

Hi,

Got 2 asterisk boxes, Box 1 has SIP users, and Box 2 has a E1 card
connected to a TDA200, when a sip user from box 1 calls someone on the
tda200 there is no ringing noise just dead silence until the person on
the TDA picks up there extensions.

Is there a way in thse situations to supply a ringing sound to the call
so the user on box 1 doesn't think there is a problem if the phone is
ringing at the other end for 20-30 seconds?

James
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RE: [Asterisk-Users] Supply ringing noise to IAX callers

[James Bean]

Whooppss after research for several hours before posting, another
asterisk user passed on the answer to me.

Add ,r to the Dial string over the E1 to hear the ringing on the line. 

James

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Bean
Sent: Monday, 11 April 2005 6:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Supply ringing noise to IAX callers


Hi,

Got 2 asterisk boxes, Box 1 has SIP users, and Box 2 has a E1 card
connected to a TDA200, when a sip user from box 1 calls someone on the
tda200 there is no ringing noise just dead silence until the person on
the TDA picks up there extensions.

Is there a way in thse situations to supply a ringing sound to the call
so the user on box 1 doesn't think there is a problem if the phone is
ringing at the other end for 20-30 seconds?

James
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RE: [Asterisk-Users] IAX calls between asterisk boxes works 1 way only

[James Bean]

Sorry again sorted it out, the [definition] has to be the same as the
username or it doesn't work, well for me anyway.

:-)

Gotta reasearch a few extra hours and play a bit more before I post I
think.

Sorry guys and girls.

James 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Bean
Sent: Monday, 11 April 2005 5:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX calls between asterisk boxes works 1 way
only


Hi,

Weird issue, 2 asterisk boxes running 1.0.7, when I call iax from box 1
to box 2 it works fine, when I dial from box 2 to box 1 I get a 

On Box 1
Apr 11 17:25:39 WARNING[8969]: chan_iax2.c:5553 socket_read: Call
rejected by 192.168.69.1: No authority found

On Box 2
Apr 11 17:26:07 NOTICE[2157]: chan_iax2.c:6573 socket_read: Rejected
connect attempt from 192.168.254.100, who was trying to reach '690@'

Error, so I obviously missed something and can someone smack me upside
the head and point out my error.

Please assume that the passwords are correct in the files :-).

Configurations are attached of each box:

Box 1

iax.conf

[general]
bindport=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
tos=lowdelay

[guest]
type=user
context=default
callerid=Guest IAX User

[salisbury]
type=friend
host=192.168.254.100
username=northbuild
secret=password
context=voip
permit=192.168.254.100

extensions.conf

[global]

PSTNLine=Zap/g1
AnalogPhone=Zap/g2

[pstn]

exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
exten = s,4,VoiceMail(u690)
exten = s,5,Hangup

[internal]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 098,1,WaitMusicOnHold(45)
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone

exten = 1690,1,VoicemailMain,s690
exten = 1691,1,VoicemailMain,s691

[outgoing]

exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup

[sip]

exten = 690,hint,SIP/snom-james
exten = 691,hint,SIP/bt-karen

exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/snom-james,30,Ttr) exten =
690,3,Voicemail2,u690 exten = 690,103,Voicemail2,b690

exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,Ttr) exten = 691,3,Voicemail,u691
exten = 691,103,Voicemail,b691

include = internal
include = outgoing
include = parkedcalls
include = voip

[voip]

exten = _1XX,1,Dial(IAX2/james:password@192.168.254.100/${EXTEN})
exten = _65X,1,Dial(IAX2/dixst:password@192.168.50.1/${EXTEN})

-
Box 2

iax.conf

[general]
bindport=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
tos=lowdelay

[guest]
type=user
context=default
callerid=Guest IAX User

[dixst]
type=friend
host=192.168.50.1
username=dixst
secret=password
context=e100p
permit=192.168.50.1

[james]
type=friend
host=192.168.69.1
username=james
secret=password
context=e100p
permit=192.168.69.1

extensions.conf

[dialstring]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

[e100p]

exten = _1XX,1,Dial(Zap/g1/${EXTEN})
exten = _93X.,1,Dial(Zap/g1/${EXTEN})
exten = _9073X.,1,Dial(Zap/g1/${EXTEN})

include = dialstring
include = voip

[voip]

exten = _65X,1,Dial(IAX2/dixst:password@192.168.50.1/${EXTEN})
; DixSt Redcliffe Ext
exten = _66X,1,Dial(IAX2/scarb:password@192.168.60.1/${EXTEN})
; Scarborough
exten = _69X,1,Dial(IAX2/northbuild:password@192.168.69.1/${EXTEN})
; James Home
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RE: [Asterisk-Users] Asterisk, Voicetronix, and Australia

[James Bean]

b) If you are planning to use SIP make sure you configure it properly
to work with NAT. 
SIP has a lot of issues with NAT. The alternative is using IAX but the
IAX deskphones 
arent as feature rich as the SIP phones. Also take into account the
cost of deploying IP phones.

Intent wrote that he is running vpn's inter-office in this case NAT is
not an issue, I do the same thing with 5 offices interconnected with
Ipsec tunnels using traffic shaping for best effect, no NAT'ing is
needed at all.

Also the grandstream phones are SIP and AIX, the new GSP-2000 is only
going to be about $180aud so is cheaper then your standard digital phone
extension and you just use the old snom220 with the extra line board, or
the alcatel which is very nice as well for reception, and if you use a
sip phone that supports AIX its not hard to set them up to fail over to
another offices asterisk box if something goes wrong, the traffic over
the vpn won't be that big with AIX, 28kbit per call if I remember right.

c) Last time I checked TDM400P wasnt A-Tick certified but things may
have changed. you 
can check with the australian suppliers
http://www.austechpartnerships.com. If you have 
4 lines this card (TDM400P) will be ideal for you.

No it still isn't certified but I have several running 4 PSTN's through,
except when running the rev h cards which you have to do a code hack to
make work (unless they fixed it in the latest zaptel), but if you can't
get it working digium support will help out.

d) Make sure your asterisk server has a decent UPS attached to it. ;)

Oohh hell yeah this is a big necessity, get one that is supported by
linux (most are these days).

f) look into agents and queues for incoming calls.

Yeah there are a few telcos out there that support IAX over the net,
they all have different ways of charging you for the calls its rather
annoying as its like comparing apples and oranges. Gotta work out what
sort of phone calls for how long you are going to do to work out which
provider to route through (it doesn't cost anything really to hook up
for more then 1 and then work out what calls to route through whom for
best cost).

Hope this helps intent.

James
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RE: [Asterisk-Users] Panasonic TDA200 E1 - E100P negotiation issues

[James Bean]

Is the E1 card an isdn card or something else? There are a several
signalling systems that can run over an E1.

When running cas you do not have a D channel for the signalling.
Instead each voice channel has a few dedicated 
bits in channel 16 (hence Channel Associated Signalling). This is used
for EM, loopstart etc and is incompatible 
with the ISDN signalling that you tried.

You need to tell us more about what card you have in the Panasonic PBX.

Ok not exactly sure what info to give you, I ordered an E1 card from
panasonic for the phone system and its what they sent me, it has an RJ45
interface and coax TX/RX connectors as well, I also have full access
with the techs version of the panasonic control/programming software
and know my way around if there is something specific I could get out of
the settings on the card for you to allow you to know which card it is.
I would assume (I know that's bad) that it is an ISDN card as it
should be the card that is used to connect to a telco directly.

I know that it is using hdb3, it only shows up with 30 channels on the
card in the E1 slot setup.

What happens to channel 16 which is usually set as the d-channel, or
should I be including channel 16 in with the rest and not using port 31
in the channels?

James
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RE: [Asterisk-Users] Panasonic TDA200 E1 - E100P negotiation issues

[James Bean]

Well, you can connect to the telco using non-isdn signalling as well.
In Europe isdn is by far 
the most common signalling form used on an E1. Can you find the model
number for the E1 card?

An E1 always has 30 voice channels, one signalling channel (running CAS
or
CCS) and one timing channel. (Well, you _can_ run voice over channel
16, but then you 
would not have any signalling as RBS is not normally used on an E1).

Channles 16 on an E1 is always reserved for signalling. There are
several signalling mechanisms 
which can be transported in that slot. Isdn uses CCS, but there are
other non-isdn signalling 
systems that instead use a few bits per channel each frame, CAS.

Can you send me the result of a pri intense debug span X from
asterisk? 
Have asterisk set to be the clock source (the timing set to 0 in the
span
line) and configured as pri_net.

Attached is the pri dump from asterisk just bringing the E1 into service
with the settings you suggested.

James
*CLI pri intense debug span 1
Enabled EXTENSIVE debugging on span 1
*CLI Sending Set Asynchronous Balanced Mode Extended

 [ 00 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 00 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 00 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 00 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 00 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 1: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 1
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 2: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 2
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 3: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 3
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 4: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 4
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 5: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 5
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 6: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 6
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 7: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 7
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 8: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 8
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 9: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 9
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 10: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 10
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 11: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 11
Mar 11 19:50:19 NOTICE[7051]: chan_zap.c:7395 pri_dchannel: PRI got event: 
Alarm (4) on Primary D-channel of span 1
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1931 pri_find_dchan: No D-channels 
available!  Using Primary on channel anyway 16!
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 12: Red Alarm
Mar 11 19:50:19 WARNING[7051]: 

RE: [Asterisk-Users] Panasonic TDA200 E1 - E100P negotiation issues

[James Bean]

Whooppss had pri_cpe set, redid the debug as attached.

They seem the same but just in case.

James
Enabled EXTENSIVE debugging on span 1
*CLI Sending Set Asynchronous Balanced Mode Extended

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 1: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 1
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 2: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 2
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 3: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 3
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 4: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 4
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 5: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 5
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 6: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 6
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 7: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 7
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 8: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 8
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 9: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 9
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 10: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 10
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 11: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 11
Mar 11 19:58:13 NOTICE[7151]: chan_zap.c:7395 pri_dchannel: PRI got event: 
Alarm (4) on Primary D-channel of span 1
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1931 pri_find_dchan: No D-channels 
available!  Using Primary on channel anyway 16!
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 12: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 12
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 13: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 13
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 14: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 14
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 15: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 15
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 17: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 

RE: [Asterisk-Users] Panasonic TDA200 E1 - E100P negotiation issues

[James Bean]

Asterisk does not see anything coming in on the D channel. What does
zttool say about the state of the link?

zttool shows the card exists, the following information shows when
select the card

Main Screen - Alarms Ok / Span Digium Wildcard E100P E1/PRA Card 0

Current Alarms: No Alarms
Sync Source: Internally clocked
IRQ Misses: 48
Bipolar Viol: 0
Tx/Rx Levels: 0/0
Total/Conf/ActL 31/31/0

Then a whole lot of numbers in a row with columns of TxA TxB etc with
dashes covering all lines.

[EMAIL PROTECTED] root]# cat /proc/interrupts
   CPU0
  0:  101341836  XT-PIC  timer
  1:526  XT-PIC  i8042
  2:  0  XT-PIC  cascade
  3:   15425613  XT-PIC  eth0
  5:  101742446  XT-PIC  t1xxp
  7:   18724522  XT-PIC  eth1
  8:  1  XT-PIC  rtc
  9:  0  XT-PIC  acpi
 11: 393426  XT-PIC  3ware Storage Controller, ohci1394
 12: 85  XT-PIC  i8042
NMI:  0
ERR:  0

As I said before, if the card is an isdn card you need to use ccs
signalling. 
Cas signalling is unusual, but possible, over an E1. Can you find out
the model 
number of the E1 card in the Panasonic pbx?

When I use cas signalling on the * server the E1 card on the TDA shows a
green sync light, when on ccs it show sync error.

Ok there was a long string of numbers on the E1 card starting right to
left at the top of the card

E1 PRI23 PRI30 PSUP1431ZB KX-TDA0187/0188/0290/0290CE

I believe the model number is a KX-TDA0290 or the 0290CE, as the 0187 is
a T1 Trunk card but we don't have T1 in australia it is E1.

I hope this is the answer to your questions sir, or did I create more?

James
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[Asterisk-Users] Panasonic TDA200 E1 - E100P negotiation issues

[James Bean]

Hi, I hope someone can help me with this
 
Asterisk 1.0.6 Zaptel 1.0.6 Libpri 1.0.6, 1 Digium E100P card installed
Panasonic TDA200 firmware v2.0.6 E1 Card Firmware 1.0.2
 
System is located in Australia, so as technologies go, I believe it is twist on 
the euro standard for the E1 signalling.
 
Here is the situation.
 
The TDA E1 card is set in cross over mode and I am using a functional standard 
straight through cable (sorry don't know the technical term, panasonic supplier 
gave me the cable premade with the card).
 
I have been systematically going through each and everyone of the span, 
signalling, crossover/patch cable settings I could find to see what would work.
 
To start if i use crc4 at any time I got no sync at either card.
 
I am pretty sure I need to supply clock with the connection as the TDA E1 is 
expecting to be plugged into Telco E1 link, but for testing I was trying every 
combination.
 
If I use span=1,x,0,ccs,hdb3 where x =0,1,2 the sync light appears on the E100P 
card but when i bring the TDA E1 card in service i get a sync error and a crap 
load of Red Alerts for every channel and a
 
Mar 11 14:40:35 NOTICE[3910]: chan_zap.c:7395 pri_dchannel: PRI got event: No 
more alarm (5) on Primary D-channel of span 1
Mar 11 14:40:35 WARNING[3910]: chan_zap.c:1931 pri_find_dchan: No D-channels 
available!  Using Primary on channel anyway 16!

At this stage I was pulling hair thinking what the but since murphy's law 
rules supreme i started thinking about other signalling types,  esf,b8zs is 
definitely not going to work so I tried.
 
span=1,x,0,cas,hdb3 and suddenly i get across the board green lights on the 
E100P and the TDA E1, but once the sync actually completes I then get the same 
red alert's on every channel including the above d-channel error. 
 
cas,hdb3 seems to be the way to go, although not generally used by what i have 
read on the net, but after sync it bombs out in asterisk, does anyone have any 
clues of what I could try next. 
 
I tried all instances swapping between pri_net and pri_cpe in the zapata.conf
 
Following is all the relevant (I think) config files.
 
[EMAIL PROTECTED] root]# cat /etc/zaptel.conf
span=1,2,0,cas,hdb3
bchan=1-15,17-31
dchan=16
loadzone=au
defaultzone=au

[EMAIL PROTECTED] root]# cat /etc/asterisk/zapata.conf
[channels]
context=default
musiconhold=default
switchtype=euroisdn
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0.0
txgain=0.0
group=1
signalling=pri_net
overlapdial=yes
context=internal
callerid=asreceived
channel=1-15, 17-31

After each change i reloaded the zaptel and wct1xxp modules and did a ztcfg -vv 
to confirm the change before restarting asterisk.
 
James
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RE: [Asterisk-Users] E1 LED not lighting up....

[James Bean]

I am no expert but I hope I can be of 
help.

What is your /etc/zaptel.conf and 
/etc/asterisk/zapata.conf?

There is 2 points it can be a problem that I have found, 
the cable, as depending on requirements a normal patch lead or a specific pin 
cross over cable maybe needed.

The other place is the span setup in the zaptel.conf and 
the signalling in the zapata.conf

Depending on where in the world you are will dictate the 
span setup for the E1.

Have a read of 

http://www.digium.com/index.php?menu=configuration#T_E100P_PRI

It helped me out with the span configuration 
requirements.

James


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Callum 
McGillivraySent: Friday, 11 March 2005 4:22 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] E1 LED not 
lighting up


Hi all,

Having an issue in getting our first 
asterisk server recognising the channels on our E1.

At the moment, when we start up the 
machine, everything goes well, but the light on our single span TE110P card 
flashes red (slowly).

I imagine that Im supposed to get a 
green light when a connection has been made. (lol  else Im not sure what its 
there for :P)

Can anyone give me some pointers as 
to what I should be looking for here? Im pretty new to this and Im eager 
to get it up and running.

Cheers,

Callum
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[Asterisk-Users] Upgraded to Asterisk 1.0.6 now crashes on boot, sql issue?

[James Bean]

I just upgraded 4 boxes to 1.0.6 without issue, then I went to upgrade
my personal test box which I am playing with call logging cdr stuff on,
writing to postgres and now asterisk now crashes on boot with the
following error.


 [app_while.so]Mar 10 17:20:04 WARNING[7239]: loader.c:258
ast_load_resource: /usr/lib/asterisk/modules/app_while.so: undefined
symbol: ast_parseable_goto
Mar 10 17:20:04 WARNING[7239]: loader.c:440 load_modules: Loading module
app_while.so failed!
[EMAIL PROTECTED] asterisk]# Warning, flexibel rate not heavily tested!
Ouch ... error while writing audio data: : Broken pipe
Ouch ... error while writing audio data: : Broken pipe

Is this related to cdr or is it to do with musiconhold?

The test box was upgraded from the 1.0.5 cvs from february where as the
other boxes were upgraded from 1.0.4 stable.

Can anyone shed some light on where I should be looking to fix it?

James
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[Asterisk-Users] CDR writing incorrect data to pgsql tables

[James Bean]

Hi,

I have postgresql and * all up and running as the latest cvs-250205,
although something weird.

Every outgoing call regardless of whether or not it is answered or busy
or just rings out in the database the entry has the disposition as
ANSWERED, instead of BUSY or NOT ANSWERED.

As a test I intentionally rang numbers that would be busy or wouldn't be
there to answer the call.

Anyone got an idea where it might be going wrong?

James
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RE: [Asterisk-Users] CDR writing incorrect data to pgsql tables

[James Bean]

 James Bean [EMAIL PROTECTED] wrote:
 [...]
  Every outgoing call regardless of whether or not it is answered or 
  busy or just rings out in the database the entry has the 
 disposition 
  as ANSWERED, instead of BUSY or NOT ANSWERED.
 
  As a test I intentionally rang numbers that would be busy 
 or wouldn't 
  be there to answer the call. Anyone got an idea where it might be 
  going wrong?
 
 Are you using analogue lines? Such lines are considered 
 answered as soon as the number has been dialled by the 
 Zaptel interface.
 
 --
 Marriage: a souvenir of love.

Yes they are analogue lines.

I am sorry I did not see anything in any of the docs about analogue
lines causing ANSWERED response on all calls.

Could you point me in the right direction to a fix or setup that fixes
this situation?

James
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RE: [Asterisk-Users] CDR writing incorrect data to pgsql tables

[James Bean]

 James Bean [EMAIL PROTECTED] wrote:
 [...]
  I am sorry I did not see anything in any of the docs about analogue 
  lines causing ANSWERED response on all calls. Could you point me in 
  the right direction to a fix or setup that fixes this situation?
 
 The only real fix is to get some form of digital service, 
 either ISDN or VoIP. There is no reliable means to detect 
 when a call has been answered on an analogue line, so 
 Asterisk doesn't bother trying.
 
 The usual kludge for analogue PBXes is to assume that a call 
 was answered only if the recorded time is longer than a 
 certain number of seconds.

Hhmm well that's annoying

Is the kludge done at the software side when the data is pulled out for
accounting and being under say 45 seconds is a no answer or busy? Or is
there a tweak that can be done at the database itself?

So by that any calls that go out over the net using IAX to the telco are
considered digital and will report correctly?

James
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RE: [Asterisk-Users] CDR writing incorrect data to pgsql tables

[James Bean]

 For MySQL and other glorified flat-file databases, you would 
 need to postprocess the data. You may feel more confident 
 skipping triggers and doing this anyway.
 
  So by that any calls that go out over the net using IAX to 
 the telco 
  are considered digital and will report correctly?
 
 Yes. You will probably be able to make the simple assumption 
 that if dstchannel ILIKE 'Zap/%' , you're going to have to 
 fudge it, otherwise it's correctly recorded.
 

Thank you for your help sir it was very informative I am going to write
the trigger with my own rules for the database and see how I go :-)

James
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RE: [Asterisk-Users] CallTransfer

[James Bean]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Mark Benson
 Sent: Thursday, 24 February 2005 8:41 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] CallTransfer
 
 I get the impression that the transfer/flash/recall etc etc 
 buttons don't always work - it seems to depend on what 
 phone/firmware you are using. And possibly the version of asterisk.
 
 I am using BT102s and some generic voip phone. On the BT102 
 the transfer button will put the call on hold and give you a 
 new line to call an extention with, however nothing happens 
 when I call an extention. On the generic voip phone the 
 transfer button does nothing.
 
 I have resorted to using # for blind xfer and *2 for attended xfer.
 

On the BT102's this got me initially to until I had a play.

When transfering a call with the BT series phone after you hit the
transfer button and it puts the call on hold and you dial the extension
you wish to transfer to you must hit the Send button for it to send the
call across. Its an attended transfer, if the other person says take a
message you can get the call back and take a message.

James
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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

 James,
 
 Are watching the SIP Messaging?  SIP Trace on the phone and 
 sip debug... on the Asterisk box?
 
 The Asterisk should be sending a NOTIFY to the Snom when that 
 hint line is hit.  If you see it in Asterisk, verify that you 
 received it on the SIP Trace page of the Snom.
 
 When rebooted, the Snom will send a SUBSCRIBE for that button 
 but Asterisk will probably not do anything with it.
 
 Have fun,
 Shanon

Thanks for the info Shanon, I do apologise I don't know 100% what I am
looking at but will give it my best shot.

I powered the snom off and on again and with sip debug enabled on * and
cleared out sip trace on the snom.

The login was pretty normal with a couple of pages of standard
negotiating going on, the snom phone in SIP Trace did notify * that it
had a hint button for the other extension (691), as attached below.

I have also attached the [sip] seciotn of extensions.conf where the
hints are.

When I went to extension 691 and dialed an external call the * box did
not send a hint/notify to the snom that 691 was in use. I checked backed
through the debug and all the logs were specifically or the call that
691 was making out zap, so the problems seems to be * nto sending the
hint to the snom phone.

Any input on this would be very much appreciated.

The one thing I have not tried is doing the hint as 

exten = 691,hint,691 ???

James Bean


Snom phone SIP Trace

NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.69.1:5060;branch=z9hG4bK72930751
From: sip:[EMAIL PROTECTED];user=phone;tag=as7d00d305
To: sip:[EMAIL PROTECTED];tag=1dz3l0jjq0
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 210

?xml version=1.0?
dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=0
state=full entity=sip:[EMAIL PROTECTED]
dialog id=691
stateterminated/state
/dialog
/dialog-info

* sip debug

Scheduling destruction of call
'[EMAIL PROTECTED]' in 361 ms
Reliably Transmitting:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.69.1:5060;branch=z9hG4bK72930751
From: sip:[EMAIL PROTECTED];user=phone;tag=as7d00d305
To: sip:[EMAIL PROTECTED];tag=1dz3l0jjq0
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 210

?xml version=1.0?
dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=0
state=full entity=sip:[EMAIL PROTECTED]
dialog id=691
stateterminated/state
/dialog
/dialog-info
 (no NAT) to 192.168.69.250:5060

---
Extensions.conf [sip] section

[sip]

exten = 690,hint,SIP/snom-james
exten = 691,hint,SIP/bt-karen

exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/snom-james,30,Ttr)
exten = 690,3,Voicemail2,u690
exten = 690,103,Voicemail2,b690

exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,Ttr)
exten = 691,3,Voicemail,u691
exten = 691,103,Voicemail,b691

include = internal
include = outgoing
include = parkedcalls

---

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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Hecken, Guido
 Sent: Thursday, 24 February 2005 3:10 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Snom phone hint exten question
 
 exten = 691,hint,SIP/691
 should do the job. I've got it working with SNOM 190 Phones 
 and actual CVS-HEAD.
 Perhaps there is a problem using the callerid instead of the 
 extension in the hint?!
 
 Hope, it helps...
 

Woohoo, it works now :-)...

Thank you very much, its weird that it requires that formating for the
extension to do the hint, but thank you none the less.

:-)

James
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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Hecken, Guido
 Sent: Thursday, 24 February 2005 3:10 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Snom phone hint exten question
 
 exten = 691,hint,SIP/691
 should do the job. I've got it working with SNOM 190 Phones 
 and actual CVS-HEAD.
 Perhaps there is a problem using the callerid instead of the 
 extension in the hint?!
 
 Hope, it helps...
 

Wooppss, I need to take that last one back.

On the snom190 the light is on solid now, whether or not the line is in
use.

Rebooting the snom doesn't change things.

I also attempted using

exten = 691,hint,691

No change.

James
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[Asterisk-Users] Grandstream 486 Sending Faxes issue out TDM400P

[James Bean]

Hi,

Hoping someone has run into the same issue.

I have an * 1.0.5 tdm400p and 2 fax machines on grandstream 486 boxes.

When a fax comes in, no problem receives it fine. When you try to send a
fax out just as the fax seems to be finishing the send you get a comms
error on the fax machine and it fails wanting to retry (tried 2
different brand fax machines same issue).

The 486's were configured to use ilbc as the codec, after looking around
I thought it might be a compression issue so I changed the 486's to alaw
with 1 TX packet per send, bandwidth not being an issue as its local
lan. Same issue.

Anyone got any ideas on what might be the issue? I used a PSTN phone on
the 486 and dialed out manually and there was no echo or noise on the
line.

James
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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

  
  I haven't used it in a while, but I had to put subscribecontext=sip 
  for the phone's (in your case the snom) sip entry.
  
  This seems like it has been removed from the wiki.  Has it 
 changed or 
  is this incorrect?
 
 
 Hi James,
 
 I have just found out that all you need to do is make the 
 hint in the context where the phone registers. That means 
 that all you need to do is put '690,hint,SIP/bt-karen' in 
 your [sip] context, nothing else and it should work. Remember 
 to take the power from the phone for a short while after you 
 have configured this, otherwise it won't work.
 
 thorben
 

Ok your example confused me a little.

You put 690,hint,SIP/bt-karen

From this section in my extensions from your example I should have

exten = 690,hint,SIP/bt-karen

exten = 691,hint,SIP/snom-james

So set hint on the opposite extensions?

[sip]

exten = 690,hint,SIP/snom-james
exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/snom-james,30,Ttr)
exten = 690,3,Voicemail2,u690
exten = 690,103,Voicemail2,b690

exten = 691,hint,SIP/bt-karen
exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,Ttr)
exten = 691,3,Voicemail,u691
exten = 691,103,Voicemail,b691
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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

 
  I am going to now sit in a corner and go quietly insane 
 while playing 
  the banyo with no strings.
  
  Still doesn't work, I dialed in an outside line and picked up the 
  receive on extension 691, yet the light on the snom phone 
 did not come 
  on. I dialed out of extension 691 to an outside line, yet still the 
  light did not come on.
  
  Snom190 has firmware 3.56m the button is set to Destination 691
 
 Hi James,
 
 I am using the latest CSV-HEAD of *, I do not think it works 
 with * stable.
 
 Thorben
 

Just downloaded the latest cvs 21/2/05 and compiled and installed it.

Still nothing, the led's work on the snom but naybe its just buggered,
*sigh*

James
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RE: [Asterisk-Users] External relay triggered by Asterisk extension-question

[James Bean]

Very friggen cool, that you very much for the information it looks like
it will do the job nicely.

What did you use in your extensions list to activate the relay?

James 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
 Sent: Sunday, 20 February 2005 6:24 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] External relay triggered by 
 Asterisk extension-question
 
 Done something similar in a different application, but * 
 should handle it --
 
 In my case, I took a crystalfontz LCD, type 633, and used two 
 of the four fan-outputs to drive two 12V relays.  As a nice 
 extra, you get temperature capabilities thrown in, so you can 
 monitor your set-up.  The LCD runs on serial, of course.
 
 As an alternative, you can use any of the many available 
 relay boards -- $50 gets you this:
 http://www.phanderson.com/iom141.html
 
  -Original Message-
  From: James Bean [mailto:[EMAIL PROTECTED]
  Sent: Saturday, February 19, 2005 11:34 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [Asterisk-Users] External relay triggered by Asterisk 
  extension -question
  
  
  
  Has anyone every setup an external open/close relay, off 
 say a serial 
  interface, and have an extension trigger the relay?
  
  Why I ask is I have a student accomodation where I am installing an 
  asterisk box to supply phone services to the tenants, there 
 is already 
  an intercom system in the main hallways that triggers the 
 downstairs 
  door and gate using a standard relay open/close trip, so I 
 was hoping 
  to get the linux box with asterisk to trip the same type of relay.
  
  Is there any door phones that are speaker driven only and sip based 
  that anyone knows about as well?
  
  James
  
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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Jon Radon
 Sent: Monday, 21 February 2005 2:55 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Snom phone hint exten question
 
 I haven't used it in a while, but I had to put 
 subscribecontext=sip for the phone's (in your case the snom) 
 sip entry.
 
 This seems like it has been removed from the wiki.  Has it 
 changed or is this incorrect?
 
 http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+ph
 one+snomdiff=7
 
 
 On Sat, 19 Feb 2005 21:36:04 +1000, James Bean 
 [EMAIL PROTECTED] wrote:
  Putting bt-karen in the destination of the snom doesn't work, i.e.
  pushing the button the phone says no such destination.
  
  exten = 691,hint,SIP/bt-karen
  exten = 691,1,SetMusicOnHold(random)
  exten = 691,2,Dial(SIP/bt-karen,30,tr) exten = 
 691,10,voicemail,u691
  
  Is in the extensions.conf but in the snom I have destination as 691.
  
  In the sip.conf it is setup as
  
  [bt-karen]
  type=friend
  secret=secret password
  host=dynamic
  callerid=Karen Colomb 691
  defaultip=192.168.69.251
  dtmfmode=info
  mailbox=691
  
  Hope this helps.
  
  James
 
 
 --
 Is it something someone said, was it something someone said?
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Thanks for the link, it had some very userful information in it,
unforunately the lights on my snom are still dead as a door nail.

Ok the snom phone has one of its LED's set to Destination 691 (it
changes that into the sip address and it dials the extension when I hit
the button on the snom no problems, and the led works)

Does anyone know where I have gone wrong.

Configurations I have enabled are voicemail and call parking.

My sip.conf is

[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = all
allow = ilbc
allow = alaw
allow = ulaw
nat=disable
srvlookup=no
localnet=192.168.69.0/255.255.255.0
subscribecontext = sip

[snom-james]
type=friend
secret=password
host=dynamic
callerid=James Bean 690
defaultip=192.168.69.250
dtmfmode=rfc2833
mailbox=690

[bt-karen]
type=friend
secret=password
host=dynamic
callerid=Karen Colomb 691
defaultip=192.168.69.251
dtmfmode=info
mailbox=691

My extensions.conf is

[pstn]

exten = s,hint,SIP/bt-karen
exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t) 
exten = s,4,VoiceMail(u690) 
exten = s,5,Hangup

[internal]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 098,1,WaitMusicOnHold(45)
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone

exten = 1690,1,VoicemailMain,s690
exten = 1691,1,VoicemailMain,s691

[outgoing]

exten = _9X.,hint,SIP/bt-karen
exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup

[sip]

exten = 690,hint,SIP/snom-james
exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/snom-james,30,Ttr)
exten = 690,3,Voicemail,u690
exten = 690,103,Voicemail,b690

exten = 691,hint,SIP/bt-karen
exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,Ttr)
exten = 691,3,Voicemail,u691
exten = 691,103,Voicemail,b691

include = internal
include = outgoing
include = parkedcalls
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[Asterisk-Users] Sip question - allow only 1 incoming call to sip phone

[James Bean]

Hi,

I need to have it so that if someone is on their sip phone that any
other attempts to contact that phone will result in a transfer to
voicemail.

Someone mentioned there might be a setting like numofcalls = 1 in the
sip.conf so that only 1 call would every be sent to the sip phone but I
would like it to goto voicemail is they are already on the phone.

Thank you in advance to all that can help.

James
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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Thorben Jensen
 Sent: Saturday, 19 February 2005 6:13 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: SV: [Asterisk-Users] Snom phone hint exten question
 
 
 
  -Oprindelig meddelelse-
  Fra: [EMAIL PROTECTED] 
 [mailto:asterisk-users- 
  [EMAIL PROTECTED] På vegne af James Bean
  Sendt: 19. februar 2005 08:14
  Til: Asterisk Users Mailing List - Non-Commercial Discussion
  Emne: [Asterisk-Users] Snom phone hint exten question
  
  
  Hi,
  
  I am sorry to be asking this but the wiki is down and has 
 been for a 
  couple of days and I need to get this working before Monday 
 to get my 
  live system setup.
  
  Trying to get the Snom 190's and soon to arrive 3com 3102's 
 to use the 
  function keys and for the life of me I can't work it out from the 
  conversations on the archive what I am going exactly wrong here?
  
  The snom 190 with function keys is extension 690, the other 
 extension
  (691) is just a BT102 so it doesn't have any function keys 
 to program.
  
  When extension 691 is dialing out, or receives a call I want it to 
  just tell the snom190 on ext 690 so the light shows up.
  
  (Soon as I got it going here I have a live system I will be 
 setting it 
  up on).
  
  Thank you to anyone in advance for the help.
  
  This is my extensions.conf
  
  ---
  
  [pstn]
  
  exten = s,1,SetMusicOnHold(random)
  exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
  exten = s,3,Hangup
  ;exten = s,10,VoiceMail(u100);Whatever box you want.
  
  [internal]
  
  exten = i,1,Playback(invalid)
  exten = i,2,Hangup
  exten = t,1,Hangup
  
  exten = 098,1,WaitMusicOnHold(5)
  exten = 099,1,Echo ;simple echo test when you dial 099 on your
  phone
  
  include = outgoing
  include = sip
  
  [outgoing]
  
  exten = _9X.,1,Dial(Zap/g1/${EXTEN:1}) exten = 
 _9X.,2,Congestion() 
  exten = _9X.,3,Hangup
  
  [sip]
  
  exten = 690,2,SetMusicOnHold(random)
  exten = 690,3,Dial(SIP/snom-james,30,tr) exten = 
  690,4,voicemail2,u690 exten = 690,102,voicemail2,b690
  
  exten = 691,hint,SIP/bt-karen
  exten = 691,1,Macro(stdexten,SIP/bt-karen)
  exten = 691,2,SetMusicOnHold(random)
  exten = 691,3,Dial(SIP/bt-karen,30,tr) exten = 
 691,4,voicemail2,u691 
  exten = 691,102,voicemail,b691
  
  include = internal
  include = outgoing
 
 Hi,
 
 You need to 'hint' SIP/bt-karen in the pstn context:
 
 [pstn]
 exten = s,hint,SIP/bt-karen
 exten = s,1,SetMusicOnHold(random)
 exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
 exten = s,3,Hangup
 
 Thorben
 
 
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Thank you for that, it was very much appreciated.

Also instead of putting a whole bunch of hints in, how might I go about putting 
a cluster of SIP extensions in the hint off the PSTN situation?

Could you also maybe throw me a couple of hints what the 

exten = 691,1,Macro(stdexten,SIP/bt-karen)

Macro portion I have seen in some examples but I am not sure what it does.

James
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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Thorben Jensen
 Sent: Saturday, 19 February 2005 6:13 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: SV: [Asterisk-Users] Snom phone hint exten question
 
 
 
  -Oprindelig meddelelse-
  Fra: [EMAIL PROTECTED] 
 [mailto:asterisk-users- 
  [EMAIL PROTECTED] På vegne af James Bean
  Sendt: 19. februar 2005 08:14
  Til: Asterisk Users Mailing List - Non-Commercial Discussion
  Emne: [Asterisk-Users] Snom phone hint exten question
  
  
  Hi,
  
  I am sorry to be asking this but the wiki is down and has 
 been for a 
  couple of days and I need to get this working before Monday 
 to get my 
  live system setup.
  
  Trying to get the Snom 190's and soon to arrive 3com 3102's 
 to use the 
  function keys and for the life of me I can't work it out from the 
  conversations on the archive what I am going exactly wrong here?
  
  The snom 190 with function keys is extension 690, the other 
 extension
  (691) is just a BT102 so it doesn't have any function keys 
 to program.
  
  When extension 691 is dialing out, or receives a call I want it to 
  just tell the snom190 on ext 690 so the light shows up.
  
  (Soon as I got it going here I have a live system I will be 
 setting it 
  up on).
  
  Thank you to anyone in advance for the help.
  
  This is my extensions.conf
  
  ---
  
  [pstn]
  
  exten = s,1,SetMusicOnHold(random)
  exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
  exten = s,3,Hangup
  ;exten = s,10,VoiceMail(u100);Whatever box you want.
  
  [internal]
  
  exten = i,1,Playback(invalid)
  exten = i,2,Hangup
  exten = t,1,Hangup
  
  exten = 098,1,WaitMusicOnHold(5)
  exten = 099,1,Echo ;simple echo test when you dial 099 on your
  phone
  
  include = outgoing
  include = sip
  
  [outgoing]
  
  exten = _9X.,1,Dial(Zap/g1/${EXTEN:1}) exten = 
 _9X.,2,Congestion() 
  exten = _9X.,3,Hangup
  
  [sip]
  
  exten = 690,2,SetMusicOnHold(random)
  exten = 690,3,Dial(SIP/snom-james,30,tr) exten = 
  690,4,voicemail2,u690 exten = 690,102,voicemail2,b690
  
  exten = 691,hint,SIP/bt-karen
  exten = 691,1,Macro(stdexten,SIP/bt-karen)
  exten = 691,2,SetMusicOnHold(random)
  exten = 691,3,Dial(SIP/bt-karen,30,tr) exten = 
 691,4,voicemail2,u691 
  exten = 691,102,voicemail,b691
  
  include = internal
  include = outgoing
 
 Hi,
 
 You need to 'hint' SIP/bt-karen in the pstn context:
 
 [pstn]
 exten = s,hint,SIP/bt-karen
 exten = s,1,SetMusicOnHold(random)
 exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
 exten = s,3,Hangup
 
 Thorben
 
 
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Unfortunately that did not work, I hard rebooted the snom phone, the bt102 and 
the asterisk server, the light just stays off, and I tested the LED on the 
button as well just to make sure its working

I also added a hint to the outgoing context so when they make an outgoing call, 
still no luck.

My extensions.conf is now

[pstn]

exten = s,hint,SIP/bt-karen
exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t) 
exten = s,3,Hangup
;exten = s,5,VoiceMail(u100);Whatever box you want.

[internal]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 098,1,WaitMusicOnHold(5)
exten = 099,1,Echo ;simple echo test when you dial 099 on your phone

Include = outgoing
include = sip

[outgoing]

exten = _9X.,hint,SIP/bt-karen
exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup

[sip]

exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/snom-james,30,tr)
exten = 690,102,voicemail2,u690

exten = 691,hint,SIP/bt-karen
exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,tr)
exten = 691,102,voicemail,u691

include = internal
include = outgoing

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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Thorben Jensen
 Sent: Saturday, 19 February 2005 8:33 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Snom phone hint exten question
 
  Unfortunately that did not work, I hard rebooted the snom 
 phone, the 
  bt102 and the asterisk server, the light just stays off, 
 and I tested 
  the LED on the button as well just to make sure its working
  
  I also added a hint to the outgoing context so when they make an 
  outgoing call, still no luck.
  
  My extensions.conf is now
  
  [pstn]
  
  exten = s,hint,SIP/bt-karen
  exten = s,1,SetMusicOnHold(random)
  exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
  exten = s,3,Hangup
  ;exten = s,5,VoiceMail(u100);Whatever box you want.
  
  [internal]
  
  exten = i,1,Playback(invalid)
  exten = i,2,Hangup
  exten = t,1,Hangup
  
  exten = 098,1,WaitMusicOnHold(5)
  exten = 099,1,Echo ;simple echo test when you dial 099 
 on your phone
  
  Include = outgoing
  include = sip
  
  [outgoing]
  
  exten = _9X.,hint,SIP/bt-karen
  exten = _9X.,1,Dial(Zap/g1/${EXTEN:1}) exten = 
 _9X.,2,Congestion() 
  exten = _9X.,3,Hangup
  
  [sip]
  
  exten = 690,1,SetMusicOnHold(random)
  exten = 690,2,Dial(SIP/snom-james,30,tr) exten = 
  690,102,voicemail2,u690
  
  exten = 691,hint,SIP/bt-karen
  exten = 691,1,SetMusicOnHold(random)
  exten = 691,2,Dial(SIP/bt-karen,30,tr) exten = 
  691,102,voicemail,u691
  
  include = internal
  include = outgoing
 
 
 Have you set the function key on the SNOM to 'Destination' 
 and typed '691'
 in the number?
 
 
 
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Ooppss sorry should have put that in, yes the snome has the function key
set to destination and 691, when I push the button it calls that
extensions.

Updated extensions.conf

[global]

PSTNLine=Zap/g1
AnalogPhone=Zap/g2

[pstn]

exten = s,hint,SIP/bt-karen
exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t) 
exten = s,3,Hangup
;exten = s,5,VoiceMail(u690) 

[internal]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 098,1,WaitMusicOnHold(45)
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone

[outgoing]

exten = _9X.,hint,SIP/bt-karen
exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup

[sip]

exten = 690,hint,SIP/snom-james
exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/snom-james,30,tr)
exten = 690,10,voicemail2,u690

exten = 691,hint,SIP/bt-karen
exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,tr)
exten = 691,10,voicemail,u691

include = internal
include = outgoing
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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Thorben Jensen
 Sent: Saturday, 19 February 2005 8:33 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Snom phone hint exten question
 
  Unfortunately that did not work, I hard rebooted the snom 
 phone, the 
  bt102 and the asterisk server, the light just stays off, 
 and I tested 
  the LED on the button as well just to make sure its working
  
  I also added a hint to the outgoing context so when they make an 
  outgoing call, still no luck.
  
  My extensions.conf is now
  
  [pstn]
  
  exten = s,hint,SIP/bt-karen
  exten = s,1,SetMusicOnHold(random)
  exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
  exten = s,3,Hangup
  ;exten = s,5,VoiceMail(u100);Whatever box you want.
  
  [internal]
  
  exten = i,1,Playback(invalid)
  exten = i,2,Hangup
  exten = t,1,Hangup
  
  exten = 098,1,WaitMusicOnHold(5)
  exten = 099,1,Echo ;simple echo test when you dial 099 
 on your phone
  
  Include = outgoing
  include = sip
  
  [outgoing]
  
  exten = _9X.,hint,SIP/bt-karen
  exten = _9X.,1,Dial(Zap/g1/${EXTEN:1}) exten = 
 _9X.,2,Congestion() 
  exten = _9X.,3,Hangup
  
  [sip]
  
  exten = 690,1,SetMusicOnHold(random)
  exten = 690,2,Dial(SIP/snom-james,30,tr) exten = 
  690,102,voicemail2,u690
  
  exten = 691,hint,SIP/bt-karen
  exten = 691,1,SetMusicOnHold(random)
  exten = 691,2,Dial(SIP/bt-karen,30,tr) exten = 
  691,102,voicemail,u691
  
  include = internal
  include = outgoing
 
 
 Have you set the function key on the SNOM to 'Destination' 
 and typed '691'
 in the number?
 
 
 
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Even worse I didn't add...

The snom firmware is the latest 3.56m

James
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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

No its setup in the snom as 691 not bt-karen I will test that now.

James 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Thorben Jensen
 Sent: Saturday, 19 February 2005 8:39 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Snom phone hint exten question
 
  
  
  Have you set the function key on the SNOM to 'Destination' 
 and typed '691'
  in the number?
 
 I am sorry, I meant that you have to type 'bt-karen' in the number.
 
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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Thorben Jensen
 Sent: Saturday, 19 February 2005 8:39 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Snom phone hint exten question
 
  
  
  Have you set the function key on the SNOM to 'Destination' 
 and typed '691'
  in the number?
 
 I am sorry, I meant that you have to type 'bt-karen' in the number.
 
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Putting bt-karen in the destination of the snom doesn't work, i.e.
pushing the button the phone says no such destination.

exten = 691,hint,SIP/bt-karen
exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,tr)
exten = 691,10,voicemail,u691

Is in the extensions.conf but in the snom I have destination as 691.

In the sip.conf it is setup as

[bt-karen]
type=friend
secret=secret password
host=dynamic
callerid=Karen Colomb 691
defaultip=192.168.69.251
dtmfmode=info
mailbox=691

Hope this helps.

James
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[Asterisk-Users] Anyone used the ACT P104SLD SIP Phone

[James Bean]

Just after some peoples impressions if they have used this phone.

It has 10 function buttons which I am hoping can be individually
programmed for destination to accept hints from asterisk.

Any input would be very much appreciated.

James
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RE: [Asterisk-Users] Anyone used the ACT P104SLD SIP Phone

[James Bean]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Joseph
 Sent: Sunday, 20 February 2005 11:35 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Anyone used the ACT P104SLD SIP Phone
 
 On Sun, 2005-02-20 at 08:38 +1000, James Bean wrote:
  Just after some peoples impressions if they have used this phone.
  
  It has 10 function buttons which I am hoping can be individually 
  programmed for destination to accept hints from asterisk.
 
 What do you mean by this?
 I'm not sure I understand.
 
 If you thing it can be used an an extension it, you are wrong. 
 Those 10 buttons for incoming calls or can be programed as 
 one touch dialing button.
 For example if you are talking on line one, and have another 
 call coming over VOIP you line two will ring etc.
 
 In general the company that make that phone doesn't even have 
 a webpage offering firmware upgrade.  So compare Sipura new 
 phone feature with this one.  At least Sipura is offering 
 constant firmware upgrade; you will most likely never see one 
 for this one.  
 
 --
 #Joseph


Sorry for not explaining myself properly. 

What I was wondering is if anyone who has used the ACT P104SLD as it has
10 function buttons, on the snom equivilent, if you program the function
button with Destination and the extension number then using hint with
asterisk it then uses the button LED on the sip phone to indicate if
that extensions is on the phone or not.

James
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[Asterisk-Users] External relay triggered by Asterisk extension - question

[James Bean]

Has anyone every setup an external open/close relay, off say a serial
interface, and have an extension trigger the relay?

Why I ask is I have a student accomodation where I am installing an
asterisk box to supply phone services to the tenants, there is already
an intercom system in the main hallways that triggers the downstairs
door and gate using a standard relay open/close trip, so I was hoping to
get the linux box with asterisk to trip the same type of relay.

Is there any door phones that are speaker driven only and sip based that
anyone knows about as well?

James

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[Asterisk-Users] MultiLine Sip Phones

[James Bean]

Sorry Newbie asking everyones option.

I am setting up a couple of small asterisk phone systems for my work, I
started using some snom 190 and bt102 sip phones (the bt102 works really
well with iLBC), but the complaint from my workmates is there is no way
to see if other people are on there phone or not, or what lines are
being used.

The snom 190 only has 5 function keys, the snom 220 seems a bit over the
top for simple users.

What suggestions do people have on some sip phones that support multiple
(6 or more but 10 or more would be better) keys where I can program
extension numbers and lines to and use hint from my asterisk box to give
updates out (I assume that's what it is for).

I was looking at the 3Com Business Phone 3102 as its not really that
expensive and looks like it comes with 18 programmable buttons which is
great, has anyone had any experience with these phones and doing this or
have any better ideas or suggestions?

As an extra note I am in Australia so not all brands are available down
here.

James Bean
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[Asterisk-Users] Snom phone hint exten question

[James Bean]

Hi, 

I am sorry to be asking this but the wiki is down and has been for a
couple of days and I need to get this working before Monday to get my
live system setup.

Trying to get the Snom 190's and soon to arrive 3com 3102's to use the
function keys and for the life of me I can't work it out from the
conversations on the archive what I am going exactly wrong here?

The snom 190 with function keys is extension 690, the other extension
(691) is just a BT102 so it doesn't have any function keys to program.

When extension 691 is dialing out, or receives a call I want it to just
tell the snom190 on ext 690 so the light shows up.

(Soon as I got it going here I have a live system I will be setting it
up on).

Thank you to anyone in advance for the help.

This is my extensions.conf

---

[pstn]

exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
exten = s,3,Hangup
;exten = s,10,VoiceMail(u100);Whatever box you want.

[internal]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 098,1,WaitMusicOnHold(5)
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone

include = outgoing
include = sip

[outgoing]

exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup

[sip]

exten = 690,2,SetMusicOnHold(random)
exten = 690,3,Dial(SIP/snom-james,30,tr)
exten = 690,4,voicemail2,u690
exten = 690,102,voicemail2,b690

exten = 691,hint,SIP/bt-karen
exten = 691,1,Macro(stdexten,SIP/bt-karen)
exten = 691,2,SetMusicOnHold(random)
exten = 691,3,Dial(SIP/bt-karen,30,tr)
exten = 691,4,voicemail2,u691
exten = 691,102,voicemail,b691

include = internal
include = outgoing

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RE: [Asterisk-Users] MultiLine Sip Phones

[James Bean]

No unfortunately a lot of the extensions do not have PC's near them or
in there offices, and the people involved are a little on the computer
illiterate side, although I am slowly training them.

They just want a phone that shows them extensions/lines and who is using
them That's why I am hoping someone else has used the 3Com Business
Phone 3102 as it comes standard with 18 function keys, just hoping they
work the same way as the snom.

James

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomas
Paseka
Sent: Saturday, 19 February 2005 5:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MultiLine Sip Phones

would an option where you could view it from a website or on your
computer be good?

coz there are a few ones out there like that, one in the wiki

James Bean wrote:

Sorry Newbie asking everyones option.

I am setting up a couple of small asterisk phone systems for my work, I

started using some snom 190 and bt102 sip phones (the bt102 works 
really well with iLBC), but the complaint from my workmates is there is

no way to see if other people are on there phone or not, or what lines 
are being used.

The snom 190 only has 5 function keys, the snom 220 seems a bit over 
the top for simple users.

What suggestions do people have on some sip phones that support 
multiple
(6 or more but 10 or more would be better) keys where I can program 
extension numbers and lines to and use hint from my asterisk box to 
give updates out (I assume that's what it is for).

I was looking at the 3Com Business Phone 3102 as its not really that 
expensive and looks like it comes with 18 programmable buttons which is

great, has anyone had any experience with these phones and doing this 
or have any better ideas or suggestions?

As an extra note I am in Australia so not all brands are available down

here.

James Bean
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[Asterisk-Users] Newbie MusicOnHold issues

[James Bean]

Hi Everyone, Merry Christmas :-)

My Asterisk Box doesn't have a sound card, it is running

Asterisk 1.02
Zaptel 1.02
Libpri 1.02
Mpg123 0.59r

All compiled from source with kernel 2.6.9-1.6 on Fedora Core 2

Any help would be very much appreciated.

The error I am getting is

-- Executing WaitMusicOnHold(SIP/snom-james-849d, 30) in new
stack
Dec 12 00:27:29 WARNING[409616]: res_musiconhold.c:366 moh1_exec: Unable
to start music on hold (class '30') on channel SIP/snom-james-849d
  == Spawn extension (sip, 098, 1) exited non-zero on
'SIP/snom-james-849d'

/etc/asterisk/musiconhold.conf
;
; Music on hold class definitions
;
[classes]
default = quietmp3:/var/lib/asterisk/mohmp3
;loud = mp3:/var/lib/asterisk/mohmp3
random = quietmp3:/var/lib/asterisk/mohmp3,-z

I also tried doing a

default = custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -z -q -r 8000
-f 8192 -b 2048 --mono -s

/etc/asterisk/extensions.conf
[pstn]

exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,2,SetMusicOnHold(random)
exten = s,3,Dial(SIP/snom-jamesSIP/bt-karen,45,t) 
exten = s,4,Hangup

[internal]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 098,1,WaitMusicOnHold(5)
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone

include = outgoing
include = sip

[outgoing]

exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup

[sip]

exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/snom-james,30,tr)
exten = 690,3,voicemail2,u690
exten = 690,102,voicemail2,b690

exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,tr)
exten = 691,3,voicemail2,u691
exten = 691,102,voicemail,b691

include = internal
include = outgoing

[from-sip]

include = internal

/etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = gsm
allow = alaw
disallow = ulaw
nat=disable
srvlookup=no
localnet=192.168.69.0/255.255.255.0
subscribecontext = sip

[snom-james]
type=friend
secret=apassword
host=dynamic
callerid=James Bean 690
defaultip=192.168.69.250
dtmfmode=rfc2833
mailbox=690

[bt-karen]
type=friend
secret=apassword
host=dynamic
callerid=Karen Colomb 691
defaultip=192.168.69.251
dtmfmode=info
mailbox=691
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RE: [Asterisk-Users] does aanyone have an example of how to dial outwith a sip phone on a pstn line?

[James Bean]

Charles S. Antrim wrote:

I am using a card that has an fxo 
and fxs module.

I am no where near 
an expert but I have my sip phone working through my pstn line and this is my 
config.

/etc/asterisk/sip.conf
[general]port = 
5060bindaddr = 192.168.69.1context = sipdisallow = gsmallow = 
alawdisallow = 
ulawnat=disablesrvlookup=nolocalnet=192.168.69.0/255.255.255.0subscribecontext 
= sip

[snom-james]type=friendsecret=passwordhost=dynamiccallerid="James 
Bean" 
690defaultip=192.168.69.250dtmfmode=rfc2833mailbox=690

[bt-karen]type=friendsecret=passwordhost=dynamiccallerid="Karen 
Colomb" 
691defaultip=192.168.69.251dtmfmode=infomailbox=691

/etc/asterisk/extensions.conf
[pstn]

exten = 
s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a comment in the CLI 
for info.exten = s,2,SetMusicOnHold(random)exten = 
s,3,Dial(SIP/snom-jamesSIP/bt-karen,45,t) exten = 
s,4,Hangup;exten = s,5,VoiceMail(u100) ;Whatever box 
you want.

[internal]

exten = 
i,1,Playback(invalid)exten = i,2,Hangupexten = 
t,1,Hangup

exten = 
099,1,Echo ;simple echo test when you dial 099 on your 
phone

include = 
outgoinginclude = sip

[outgoing]

exten = 
_9X.,1,Dial(Zap/g1/${EXTEN:1})exten = _9X.,2,Congestion()exten = 
_9X.,3,Hangup

include = 
sip

[sip]

exten = 
690,1,SetMusicOnHold(random)exten = 
690,2,Dial(SIP/snom-james,30,tr)exten = 690,3,voicemail2,u690exten 
= 690,102,voicemail2,b690

exten = 
691,1,SetMusicOnHold(random)exten = 
691,2,Dial(SIP/bt-karen,30,tr)exten = 691,3,voicemail2,u691exten 
= 691,102,voicemail,b691

include = 
internalinclude = outgoing

[from-sip]

include = 
internal

This isn't the best example of how to do it but it 
works.

I hope it helps.

James

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[Asterisk-Users] Udev setup question for zaptel

[James Bean]

Trying to setup asterisk and zaptel on a Fedora Core 3. Its all working
after reading up on udev but I still get errors.

[EMAIL PROTECTED] ~]# ztcfg -v

Zaptel Configuration
==


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

2 channels configured.

Notice: Configuration file is /etc/zaptel.conf
line 4: Unable to open master device '/dev/zap/ctl'

I added the suggested lines to /etc/udev/rules.d/50-udev.rules that were
in the zaptel README.udev, as I understood them?

# Section for zaptel device
KERNEL=zapctl, NAME=zap/ctl
KERNEL=zaptimer,   NAME=zap/timer
KERNEL=zapchannel, NAME=zap/channel
KERNEL=zappseudo,  NAME=zap/pseudo
KERNEL=zap[0-9]*,  NAME=zap/%n

When I load the zaptel modules, they work the errors are just
distracting.

Any suggestions would be great.

James
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RE: [Asterisk-Users] SNOM 190 Dial-Plan String Settings

[James Bean]

I don't have any soft phones setup, the SNOM receives the calls no problems when the 
SNOM tries to dial out it says Not Found: on the phone display, on the asterisk 
console with asterisk -vgc when I try to dialout I only get

chan_sip.c:7561 handle_request: Unknown SIP command 'PUBLISH' from '192.168.69.250'

Which I am told is normal as asterisk doesn't support the Publish command

I have an analog phone plugged into the TDM400P in another port and it dials out 
without issue.

Thanks for the response this has been bugging the crap out of me, any help would be 
appreciated.

My /etc/extensions.conf is as follows.

[pstn]

exten = s,1,Wait(2)
exten = s,2,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a comment in the 
CLI for info.
exten = s,3,Dial(SIP/snom-james,45,t)  ;Dial James SNOM Phone for incoming calls
exten = s,4,Hangup

[internal]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 099,1,Echo ;simple echo test when you dial 099 on your phone

include = outgoing
include = voip
include = sip

[outgoing]

exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup

[voip]

exten = _1XX,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM}) ; 1xx extension to 
Salisbury
exten = _2XX,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})  ; 2xx extension to 
Marcoola
exten = 610,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})  ; 610 to Jindalee
exten = 620,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})  ; 620 to Batteryhill

[sip]

exten = 690,1,Dial(SIP/snom-james,30,tr)
exten = 690,2,voicemail2,u900
exten = 690,102,voicemail2,b900

exten = 691,1,Dial(SIP/bt-karen,30,tr)
exten = 691,2,voicemail2,u901
exten = 691,102,voicemail,b901

My sip.conf is as follows

[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = gsm
allow = alaw
disallow = ulaw
nat=disable
srvlookup=no
localnet=192.168.69.0/255.255.255.0
subscribecontext = sip

[snom-james]
type=friend
secret=password deleted
host=dynamic
callerid=James 690
defaultip=192.168.69.250
dtmfmode=rfc2833
mailbox=900

[bt-karen]
type=friend
secret=password deleted
host=dynamic
callerid=Karen 691
defaultip=192.168.69.251
dtmfmode=rfc2833
mailbox=901

---

Although when I first start asterisk up I always get this 1 error that I am not sure 
about.

chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for 
seqno 102 (Non-critical Request)

James

Hello James,

There is nothing special with the Snom phones. The empty dialplan string 
is normal. You only have to specify the displayname, account, password 
and registrar. I think you have a mistake in your extensions.conf. Does 
it work with another (soft)phone?

Regards,
Joris



On Oct 15, 2004, at 1:51 PM, James Bean wrote:

 I am having a problem with my new SNOM190 and my asterisk box.
  
 Incoming calls to the SNOM work perfectly, but when i dial-out I get a 
 Not Found: number dialed on the SNOM display everytime I try, 
 nothing shows up on the console of the asterisk box so its not even 
 touching it.
  
 I have the latest 3.54 firmware on it and when I looked at the Line 1 
 setup for my asterisk box I released that in the SNOM phone there is 
 nothing in my Dial-Plan String I take it it matches this inside the 
 phone to choose which line to use in the SNOM phone.
  
 Unfortunately I am not finding much on the format of the Dial-Plan 
 String in the SNOM phones.
  
 All I need is for it to send all calls regardless of format to the 
 asterisk box.
  
 Anyone got any suggestions.
  
 James


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[Asterisk-Users] OH323 VoIP router connect debug question?

[James Bean]

Hi,

I do apologise I only have a basic understanding of VoIP and H323, here
is my situation, any help would be very much appreciated.

I am trying to coax my asterisk 1.0.1 box using oh323 0.6.3b with
openh323 13.5  pwlib v1.6.6 (I purchased 1 G.729 license from digium
and installed it correctly) to communicate with a clients OKI BV1250 (I
know cheap ass VoIP Gateway, but they have 2 and I gotta get asterisk to
talk with them).

When I was first testing it with the old version with h323 there was a
h.323 trace mode which was easy to tell why it wasn't working.

Unfortunately I can't seem to find the same functionality with OH323.

What I am getting when I make a call to the Oki VoIP box is 

-- Executing Dial(SIP/snom-james-02d1,
OH323/[EMAIL PROTECTED]/690) in new stack
-- H.323 call to [EMAIL PROTECTED]/690 with codec ALAW
-- Called [EMAIL PROTECTED]/690
-- H.323 call 'ip$localhost/19967' cleared, reason 9 (Connection
failure)
-- Hungup 'OH323/L19967'
  == No one is available to answer at this time

Sometimes the error comes up immediately, sometimes it takes 10 seconds
before you see it.

My /etc/asterisk/oh323.conf is

[general]
listenAddress=192.168.69.1
listenPort=1720
connectPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=no
h245Tunnelling=no
h245inSetup=no
inBandDTMF=no
silenceSuppression=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=2
inboundMax=2
simultaneousMax=2
bandwidthLimit=12
wrapLibTraceLevel=1
libTraceLevel=0
libTraceFile=stdout
gatekeeper=DISCOVER
gatekeeperTTL=600
userInputMode=TONE
amaFlags=default
accountCode=H323
context=voip-h323

[register]
alias=asterisk
alias=123
context=all-aliases
alias=ASTERISK
alias=666
context=more-aliases
alias=665
context=all-prefixes
gwprefix=00
gwprefix=01
context=more-stuff
alias=664
gwprefix=02

[codecs]
codec=G711A
frames=20

In my extensions.conf in my voip section I have

exten = _1XX,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})
; 1xx extension to Salisbury
exten = _2XX,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})
; 2xx extension to Marcoola
exten = 610,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})  ; 610
to Jindalee
exten = 620,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})  ; 620
to Batteryhill



james
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[Asterisk-Users] SNOM 190 Dial-Plan String Settings

[James Bean]

I am having a problem with my new SNOM190 and my asterisk 
box.

Incoming calls to the SNOM work perfectly, but when i 
dial-out I get a "Not Found: number dialed" on the SNOM display 
everytime I try, nothing shows up on the console of the asterisk box so its not 
even touching it.

I have the latest 3.54 firmware on it and when I looked at 
the Line 1 setup for my asterisk box I released that in the SNOM phone there is 
nothing in my "Dial-Plan String" I take it it matches this inside the phone to 
choose which line to use in the SNOM phone.

Unfortunately I am not finding much on the format of the 
Dial-Plan String in the SNOM phones.

All I need is for it to send all calls regardless of format 
to the asterisk box.

Anyone got any suggestions.

James
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RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P

[James Bean]

Yeah I have callerid=asreceived in my zapata.conf still nothing
unfortunately.

James 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Loftis
Sent: Wednesday, 13 October 2004 4:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P



--On Wednesday, October 13, 2004 16:04 +1000 James Bean
[EMAIL PROTECTED]
wrote:

 a) Ensure you actually have the callerid service provided to your 
 line,
 this is usually an extra charge from telstra (AFAIK)

 Yep my analog handset on the line (not through asterisk) displays the 
 callerid of the incoming call (just as a double check).

I might be wrong here, but don'y you also need callerid=asreceived on
the incoming Zap channel in zapata.conf as well?
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RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P

[James Bean]

Its getting pretty well spread here with several ISP's/Telco's offering
IAX connectivity for cheap calls.

It's growing, I hope we can just sort out the callerid thing :-).

Although I could name the line it comes in on so it doesn't just say
asterisk when the call comes in.

James 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Wednesday, 13 October 2004 4:42 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P


James - I have the same problem, and tried a some of the same ideas. No
result.

But at least we both know that a few people in Australia are using
Asterisk!

Later,

PaulH
 

-Original Message-
From: James Bean [mailto:[EMAIL PROTECTED]
Sent: Wednesday, 13 October 2004 4:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P

 Sorry, I explained this wrong.
 
 I am wanting the callerid of the incoming caller from my analogue
line 
 on the TDM400P to be passed TO the sip phone so the sip phone display

 shows the phone number of the incoming caler from the call on the
 TDM400P.
 
 It shows any callerid information from other sip phones or extension 
 calls fine.

I'm not sure, but try the following:

a) Ensure you actually have the callerid service provided to your line,
this is usually an extra charge from telstra (AFAIK)

Yep my analog handset on the line (not through asterisk) displays the
callerid of the incoming call (just as a double check).

b) Add a wait(3) (maybe a 2 or 4 seconds is needed) before the noop

Took it out to Wait(5), and made sure that the callerid was being
displayed on my analog handset before the wait times out in asterisk to
do the noop.
Still no go.

SIP handset still displays Asterisk on it when the call is patched
through.

c) Patch asterisk with this patch (I'm still waiting to be able to do
this from a config file. This is what I use to allow asterisk to pass
callerid *to* my analog FXS extensions. I assume it is the same for FXO
lines.

diff -ur asterisk/channels/chan_zap.c asterisk.mine/channels/chan_zap.c
--- asterisk/channels/chan_zap.cWed Sep 22 18:24:18 2004
+++ asterisk.mine/channels/chan_zap.c   Wed Sep 22 18:24:41 2004
@@ -89,7 +89,7 @@
 /* #define ZAP_CHECK_HOOKSTATE */

 /* Typically, how many rings before we should send Caller*ID */
-#define DEFAULT_CIDRINGS 1
+#define DEFAULT_CIDRINGS 2

 #define CHANNEL_PSEUDO -12

Obviously after the last one, you need to re-compile and re-install
asterisk, and then re-start asterisk.

Regards,
Adam

Yes I had found this patch previous and it was already compiled into my
current build, asterisk 1.0.1...

Thanks for the reply though it did open my eyes to a few things.

Unfortunately no callerid from the incoming analog line call on my
TDM400P.

James
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[Asterisk-Users] Dialing out with SIP phone problem

[James Bean]

I am trying to setup a SNOM 190 with my asterisk box but having a few
problems

When a call comes in it connects and rings and I can talk no problems...

If I try to call out with the phone I get...

NOTICE[-165364816]: chan_sip.c:7561 handle_request: Unknown SIP command
'PUBLISH' from '192.168.69.250'

I know dialing out works correctly from my analog phone plugged into my
TDM400P but the sip phone doesn't seem to dial properly?

I updated the latest firmware on the snom190...

The configuration on the SNOM190 is pretty standard with just Line 1
configured for asterisk with the correct password etc, I get the 

-- Saved useragent snom190-3.54 for peer snom-james
And
[2]24/12/2001 11:00:09: Registered at registrar as
[EMAIL PROTECTED]

So the phone and asterisk sync and talk ok.


/etc/asterisk/sip.conf

[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = gsm
allow = alaw
disallow = ulaw
srvlookup=no

[snom-james]
type=friend
secret=password removed
host=dynamic
callerid=James 690
defaultip=192.168.69.250
dtmfmode=rfc2833
mailbox=900

[bt-karen]
type=friend
secret=password removed
host=dynamic
callerid=Karen 691
defaultip=192.168.69.251
dtmfmode=rfc2833
mailbox=901

/etc/asterisk/extension.conf

[pstn]

exten = s,1,Wait(2)
exten = s,2,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,3,Dial(SIP/snom-james,45,t)  ;Dial the group=1 zap card mod
above
exten = s,4,Hangup
;exten = s,5,VoiceMail(u100);Whatever box you want.

[internal]

exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone

include = outgoing
include = voip
include = sip

[outgoing]

exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup

[voip]

exten = _1XX,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})
; 1xx extension to Salisbury
exten = _2XX,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})
; 2xx extension to Marcoola
exten = 610,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})  ; 610
to Jindalee
exten = 620,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})  ; 620
to Batteryhill

;exten = _54XX,1,Dial(OH323/[EMAIL PROTECTED]) ; 54 to
Marcoola
;exten = _0754XX,1,Dial(OH323/[EMAIL PROTECTED]); 54 to
Marcoola

[sip]

exten = 690,1,Dial(SIP/snom-james,30,tr)
exten = 690,2,voicemail2,u900
exten = 690,102,voicemail2,b900

exten = 691,1,Dial(SIP/bt-karen,30,tr)
exten = 691,2,voicemail2,u901
exten = 691,102,voicemail,b901

-

Although something strange, on bootup asterisk console displays

WARNING[-165811280]: chan_sip.c:681 retrans_pkt: Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
102 (Non-critical Request)

Any help would be very much appreciated.

James
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[Asterisk-Users] Passing CallerID to SIP phone from TDM400P

[James Bean]

Title: Passing CallerID to SIP phone from TDM400P







Hi,


Sorry, newbie, I want to pass the incoming callerid information through to my sip phone but when an incoming call gets passed through it says asterisk on the display instead of the number.

Being in australia callerid information is passed through on the second ring not the first, (hence my noop command doesn't currently work)

James


--


/etc/asterisk/extensions.conf


[pstn]


exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a comment in the CLI for info.

exten = s,2,Dial(SIP/snom-james,45,t)

exten = s,3,Hangup

;exten = s,3,VoiceMail(u100) ;Whatever box you want.


[internal]


exten = i,1,Playback(invalid)

exten = i,2,Hangup

exten = t,1,Hangup


exten = 099,1,Echo ;simple echo test when you dial 099 on your phone


include = sip


[sip]


exten = 690,1,Dial(SIP/snom-james,30,tr)

exten = 690,2,voicemail2,u900

exten = 690,102,voicemail2,b900


exten = 691,1,Dial(SIP/bt-karen,30,tr)

exten = 691,2,voicemail2,u901

exten = 691,102,voicemail,b901

 


/etc/asterisk/sip.conf


[general]


port = 5060

bindaddr = 192.168.69.1

context = sip

disallow = gsm

allow = alaw

disallow = ulaw

srvlookup=no


[snom-james]

type=friend

secret=password removed

host=dynamic

callerid=James 690

defaultip=192.168.69.250

dtmfmode=inband

mailbox=690


[bt-karen]

type=friend

secret=password removed

host=dynamic

callerid=Karen 691

defaultip=192.168.69.251

dtmfmode=inband

mailbox=691



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RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P

[James Bean]

Sorry, I explained this wrong.

I am wanting the callerid of the incoming caller from my analogue line on the TDM400P 
to be passed TO the sip phone so the sip phone display shows the phone number of the 
incoming caler from the call on the TDM400P.

It shows any callerid information from other sip phones or extension calls fine.

James 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Emilio Panighetti
Sent: Wednesday, 13 October 2004 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P

If the extension is a SIP Phone, it's up to the SIP Phone to pass the CallerID 
information. Some ATAs allow you to configure how's the Caller_ID being transmitted 
(like Cisco ATA-186). Others don't.

if you call from the console, the Caller ID information will say 'asterisk'. from your 
phones, it won't.

If the call originates, for example, from a SIP endpoint (phone, etc). 
it uses the callerid defined on sip.conf.

In your example, take the double quotes off (that seems to work in my
case):

 [bt-karen]
  type=friend
  secret=password removed
  host=dynamic
  callerid=Karen 691
  defaultip=192.168.69.251
 dtmfmode=inband
 mailbox=691

That would be what I would do.

On Oct 13, 2004, at 12:38 AM, James Bean wrote:



 Hi,

 Sorry, newbie, I want to pass the incoming callerid information 
 through to my sip phone but when an incoming call gets passed through 
 it says asterisk on the display instead of the number.

 Being in australia callerid information is passed through on the 
 second ring not the first, (hence my noop command doesn't currently
 work)

 James

 --

 /etc/asterisk/extensions.conf

 [pstn]

 exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a 
 comment in the CLI for info.
  exten = s,2,Dial(SIP/snom-james,45,t)  exten = s,3,Hangup  ;exten 
 = s,3,VoiceMail(u100)    ;Whatever box you want.

 [internal]

 exten = i,1,Playback(invalid)
 exten = i,2,Hangup
  exten = t,1,Hangup

 exten = 099,1,Echo ;simple echo test when you dial 099 on your 
 phone

 include = sip

 [sip]

 exten = 690,1,Dial(SIP/snom-james,30,tr)  exten = 
 690,2,voicemail2,u900 exten = 690,102,voicemail2,b900

 exten = 691,1,Dial(SIP/bt-karen,30,tr)  exten = 
 691,2,voicemail2,u901 exten = 691,102,voicemail,b901
  

  /etc/asterisk/sip.conf

 [general]

 port = 5060
  bindaddr = 192.168.69.1
  context = sip
  disallow = gsm
 allow = alaw
  disallow = ulaw
 srvlookup=no

 [snom-james]
  type=friend
  secret=password removed
  host=dynamic
  callerid=James 690
  defaultip=192.168.69.250
 dtmfmode=inband
 mailbox=690

 [bt-karen]
  type=friend
  secret=password removed
  host=dynamic
  callerid=Karen 691
  defaultip=192.168.69.251
 dtmfmode=inband
 mailbox=691
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RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P

[James Bean]

 Sorry, I explained this wrong.
 
 I am wanting the callerid of the incoming caller from my analogue
line 
 on the TDM400P to be passed TO the sip phone so the sip phone display

 shows the phone number of the incoming caler from the call on the 
 TDM400P.
 
 It shows any callerid information from other sip phones or extension 
 calls fine.

I'm not sure, but try the following:

a) Ensure you actually have the callerid service provided to your line,
this is usually an extra charge from telstra (AFAIK)

Yep my analog handset on the line (not through asterisk) displays the
callerid of the incoming call (just as a double check).

b) Add a wait(3) (maybe a 2 or 4 seconds is needed) before the noop

Took it out to Wait(5), and made sure that the callerid was being
displayed on my analog handset before the wait times out in asterisk to
do the noop. Still no go.

SIP handset still displays Asterisk on it when the call is patched
through.

c) Patch asterisk with this patch (I'm still waiting to be able to do
this from a config file. This is what I use to allow asterisk to pass
callerid *to* my analog FXS extensions. I assume it is the same for FXO
lines.

diff -ur asterisk/channels/chan_zap.c asterisk.mine/channels/chan_zap.c
--- asterisk/channels/chan_zap.cWed Sep 22 18:24:18 2004
+++ asterisk.mine/channels/chan_zap.c   Wed Sep 22 18:24:41 2004
@@ -89,7 +89,7 @@
 /* #define ZAP_CHECK_HOOKSTATE */

 /* Typically, how many rings before we should send Caller*ID */
-#define DEFAULT_CIDRINGS 1
+#define DEFAULT_CIDRINGS 2

 #define CHANNEL_PSEUDO -12

Obviously after the last one, you need to re-compile and re-install
asterisk, and then re-start asterisk.

Regards,
Adam

Yes I had found this patch previous and it was already compiled into my
current build, asterisk 1.0.1...

Thanks for the reply though it did open my eyes to a few things.

Unfortunately no callerid from the incoming analog line call on my
TDM400P.

James
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[Asterisk-Users] Digits being dropping when dialing from certain analog phones

[James Bean]

FC2, Asterisk 1.0.0, Zaptel 1.0.0
TDM400P Port 1 FXS Port 4 FXO

Standard analogue handset plugged in with pstn line.

Problem:

I have 2 analog phones that I use, when plugged directly into pstn line
both phones work perfectly, dialing no issues. When I plug the handsets
into the TDM400P, one works perfectly the other drops random numbers.
Its like the tone is slightly different on the second handset and its
not picking up some numbers (12356 it seems). Is there a way to adjust
the tone detection, make it more sensitive?

Keys dialed from handset were

9 0418800185

I tried hitting the keys slowly as well as at my normal speed, all tones
are heard in the handset for all numbers.



Error in asterisk -vvvgc

-- Starting simple switch on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/g2/088008) in new stack
-- Called g2/088008
-- Zap/4-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/4-1
-- Hungup 'Zap/4-1'
  == Spawn extension (internal, 9088008, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/g2/0488008) in new stack
-- Called g2/0488008
-- Zap/4-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/4-1
-- Hungup 'Zap/4-1'
  == Spawn extension (internal, 90488008, 1) exited non-zero on
'Zap/1-1'
-- Hungup 'Zap/1-1'



/etc/zaptel.conf

fxols=1
fxsls=4
Loadzone=au

/etc/zapata.conf

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxo_ls
callgroup=1
pickupgroup=1
immediate=no
context=internal
busydetect=yes
callerid=James Bean690  ;assuming extension 690
mailbox=690 ;stutter tone for voicemail - you can
use an optional context here
transfer=yes
channel=1
group=2
signalling=fxs_ls
context=pstn
channel=4

/etc/asterisk/extensions.conf

[pstn]

exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,2,Dial(Zap/g1,45,t)  ;Dial the group=1 zap card mod above
exten = s,3,Hangup

[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone

exten = _9X.,1,Dial(Zap/g2/${EXTEN:1})
exten = _9X.,2,Congestion()


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[Asterisk-Users] TDM400P Newbie configuration hell :-)

[James Bean]

Sorry to post such a newb set of questions but I have been hammering
about trying to get Asterisk running on FC2 machine reading everything
available (I think that is what stuffed me, shouldn't have read it all
:-) ).

Config

FC2 running Asterisk 1.0.0, with the h323 compiled in and installed
correctly.

Amazingly enough I have everything compiled correctly and installed.

I am running a TDM400P, Port 1 FXS, Port 4 FXO.

I have my PSTN line plugged into 1 port and my Analogue phone plugged
into port 4 (I think that's right I get tone on the phone when I pick it
up and echo works).

/etc/zaptel.conf

fxols=1
fxsls=4
; Weird but I was told to have the fxols fxsls reverse to the actually
module
loadzone = au
defaultzone = au

/etc/zapata.conf

[channels]
context=default
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxo_ls
callgroup=1
pickupgroup=1
immediate=no
context=internal
busydetect=yes
callerid=James Bean690  ;assuming extension 690
mailbox=690 ;stutter tone for voicemail - you can
use an optional context here
transfer=yes
channel=1
group=2
signalling=fxs_ls
context=pstn
channel=4

Extensions.conf

[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone

[outgoing]

exten = _1XX,1,Dial(H323/[EMAIL PROTECTED]) ; 1xx extension
to Salisbury
exten = _2XX,1,Dial(H323/[EMAIL PROTECTED])  ; 2xx extension
to Marcoola
exten = 610,1,Dial(H323/[EMAIL PROTECTED])  ; 610 to Jindalee
exten = 620,1,Dial(H323/[EMAIL PROTECTED])  ; 620 to Batteryhill

exten = _54XX,1,Dial(H323/[EMAIL PROTECTED]) ; 54 to Marcoola
exten = _0754XX,1,Dial(H323/[EMAIL PROTECTED]); 54 to
Marcoola

exten = _,1,Dial(Zap/g2/${EXTEN})

H323.conf

[general]
port = 1720
bindaddr = 192.168.69.1 
tos=lowdelay

disallow=all
allow=g723.1
allow=gsm

--

I can pick up the phone and ring 099 and echo works but if I dial
anything else I just get a busy signal with no errors on asterisk
-c, what I need is for ANY incoming calls to make the analogue phone
ring.

Outgoing calls that fit the rules use h323, everything else should pick
up the PSTN line and dial.

I again apologise for the mess and newbness (did I just invent a word),
I just need a kick start and get the basic stuff working before I start
playing.

Also, anyone had asterisk talking to OKI Voip like BV1250 units
working?, if so can you drop me an email.

James
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[Asterisk-Users] Help with dialing out with TDM400P

[James Bean]

Scenario, 

I got some very good help earlier from Joseph getting me up and started
but I have a couple of small problems still.

Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4

Analog dialout line and Analog handset plugged in.

Problems:

1.
Incoming calls work and the phone rings and can be answered no problems,
(although I wouldn't mind being able to adjust the ring but that's not
important), I can't ring out, I just get a busy signal and nothing comes
up on the console. I am pretty sure its just a simple line missing from
extensions.conf.

2.

I am based in australia and when I have an incoming call with callerid
turned on then I get the following error on console.

-- Zap/1-1 is ringing
Sep 25 22:49:14 WARNING[-203428944]: chan_zap.c:3413 zt_handle_event:
Didn't finish Caller-ID spill.  Cancelling.

---

/etc/zaptel.conf

fxols=1
fxsls=4
loadzone=au

/etc/asterisk/extensions.conf

[pstn]

exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,2,Dial(Zap/g1,45,t)  ;Dial the group=1 zap card mod above
#exten = s,3,VoiceMail(u100);Whatever box you want.

[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 099,1,Echo ;simple echo test

/etc/asterisk/zapata.conf

[channels]
context=default
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxo_ls
callgroup=1
pickupgroup=1
immediate=no
context=internal
busydetect=yes
callerid=James Bean690  ;assuming extension 690
mailbox=690 ;stutter tone for voicemail - you can
use an optional context here
transfer=yes
channel=1
group=2
signalling=fxs_ls
context=pstn
channel=4

---

Any help would be very much appreciated.

James
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[Asterisk-Users] Dropping numbers on dialout through tdm400p

[James Bean]

Specs

FC2, Asterisk 1.0.0, Zaptel 1.0.0
TDM400P Port 1 FXS Port 4 FXO

Standard analogue handset plugged in with pstn line.

Problem:

When I go to dialout it drops numbers on the outgoing number.

Keys dialed from handset were

9 0418800185

I tried hitting the keys slowly as well as at my normal speed, all tones
are heard in the handset for all numbers.



Error in asterisk -vvvgc

-- Starting simple switch on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/g2/088008) in new stack
-- Called g2/088008
-- Zap/4-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/4-1
-- Hungup 'Zap/4-1'
  == Spawn extension (internal, 9088008, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/g2/0488008) in new stack
-- Called g2/0488008
-- Zap/4-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/4-1
-- Hungup 'Zap/4-1'
  == Spawn extension (internal, 90488008, 1) exited non-zero on
'Zap/1-1'
-- Hungup 'Zap/1-1'



/etc/zaptel.conf

fxols=1
fxsls=4
Loadzone=au

/etc/zapata.conf

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxo_ls
callgroup=1
pickupgroup=1
immediate=no
context=internal
busydetect=yes
callerid=James Bean690  ;assuming extension 690
mailbox=690 ;stutter tone for voicemail - you can
use an optional context here
transfer=yes
channel=1
group=2
signalling=fxs_ls
context=pstn
channel=4

/etc/asterisk/extensions.conf

[pstn]

exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,2,Dial(Zap/g1,45,t)  ;Dial the group=1 zap card mod above
exten = s,3,Hangup

[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone

exten = _9X.,1,Dial(Zap/g2/${EXTEN:1})
exten = _9X.,2,Congestion()

--

Secondary issue, when an incoming call into the asterisk box arrives on
the asterisk terminal it shows callerid of the caller as 690 which is
the extension number that rings not the actual other persons caller id.

James
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[Asterisk-Users] zaptel.conf question

[James Bean]

Sorry very very very newbie here,

I just started setting up a asterix box as a test environment for my
work to see if it is a viable solution.

I have a standard TMD400P Development Kit with a FXS and FXO module on
it, and a standard analog handset plugged into the FXS module and a
Analog phone line plugged into a FXO.

My hope is to setup asterix to communicate with an existing OKI VoIP
network. No NAT required, all communication is by dedicated secured VPN.

Sorry for my lack of knowledge in this area but if someone could point
me in the right direction or send me a zaptel.conf and zaptela.conf that
would work in my situation it would be very much appreciated, some of
the basic text files I am finding on the net seem a little
contradictory.

James
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