--- In asterisk-anz@yahoogroups.com, "jjones_1st" <b.russell@...>
It should not a problem, make sure that your cel dialling rights are
there. is inbound working fine? as you said local dialing is working
fine,so cell also should work fine.
wrote:
>
> Hi,
>
> I have an issue where I don't seem to be able to dial mobile
phones
> from my Primary Rate service supplied by telstra.
>
> I can dial any local numbers using 9 digits (73001xxxx) interstate
I
> have to dial 39415xxxx (no leading zero) and for mobiles I have
tried
> 0417xxxxxx, 417xxxxxx, neither of which worked, just get back a
> "Channel 1/1, span 1 got hangup request, cause 31".
>
> Telstra claim that it is my equipment as the service is fully
open,
> so I am looking for some hints as to what I need to do to debug
the
> issue. I have googled around but not really found anything useful.
>
> So I guess I might have 2 issues that might be related.
> 1. only having to dial 39415xxxx instead of 03 9415 xxxx
> 2. not being able to dial mobiles at all.
>
> Any help greatly appreciated.
> My environment is :-
> Digium TE110p (the card I got on Bootcamp)
> Centos 4.6
> asterisk-1.4.18
> zaptel-1.4.9
> libpri-1.4.3
>
> Zaptel.conf
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15
> dchan=16
> bchan=17-31
> loadzone = au
> defaultzone=au
>
>
> Zapata.conf
> ;
> ; Zapata telephony interface
> ;
> ; Configuration file
> ;
> ; You need to restart Asterisk to re-configure the Zap channel
> ; CLI> reload chan_zap.so
> ;will reload the configuration file,
> ;but not all configuration options are
> ; re-configured during a reload.
>
>
>
> [trunkgroups]
> trunkgroup => 1,16
> spanmap => 1,1,1
>
> [channels]
>
> context=inside
>
> ; Setup for the Primary rate card
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=yes
> switchtype=EUROISDN
> context=from_outside
> signalling=pri_cpe
> group=3
> channel => 1-15
> channel => 17-31
> callerid=asrecieved
> pridialplan=unknown
> prilocaldialplan=unknown
> overlapdial=yes
>
> signalling=fxo_ls
> rxwink=300; Atlas seems to use long (250ms) winks
> hidecallerid=no
> callwaiting=yes
> restrictcid=no
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> callgroup=1
> pickupgroup=1
>
> immediate=no
>
> Relevant part of the dial plan
> exten => _xxxxxxxxxx,1,noop(${EXTEN} --- ${CALLERID(num)})
> exten => _xxxxxxxxxx,n,set(CALLERID(num)=73231${CALLERID(num)})
> exten => _xxxxxxxxxx,n,dial(zap/g3/${EXTEN:-9})
> exten => _[345]xxxxxxx,1,noop(${EXTEN} --- ${CALLERID(num)})
> exten => _[345]xxxxxxx,n,set(CALLERID(num)=73231${CALLERID(num)})
> exten => _[345]xxxxxxx,n,dial(zap/g3/7${EXTEN})
>

--- In asterisk-anz@yahoogroups.com, "ambreen SHEIKH" <ANBREEN@...>
HI,
You dont need to use sip for file transferring bec. SIP is not
developed for this purpose its used for RTP. Yes the one way is that
you can use linux utility SCP in asterisk dial plan, like do /system
calls or/exec funtion and run scp -f source file or files
root@ip:/path

you can search on VOIP-INFO regarding this.if you still need any
help just give me a buzz.no problem.

Regards,
Jehanzaib Younis
ph: +1484-687-5558
cell(pak):+92-321-5139853

wrote:
>
> hello everybody
>
>
>
> i have a query regarding regarding file transfer using asterisk
server
> , first of all i am not even sure if this is possible ....
>
> secondly if this is possible then i want to know what kind of
client
> will we be using to communicate with the asterisk server in order
to
> transmit a file.
>
> like for example mostly we are using the sip phones as our
> clients to communicate with the server ...... kindly help me in
this
> regard that what kind of clients do we require for file transfer.
>
> i dont think sip phones can be used to transfer a file ......
>
> (maybe we dont need any client ..... but if we dont need a client
then
> what will we use to communicate with the server)
>
>
> i will be glad if someone can guide me in this regard.
>
>
> Ambreen Sheikh
>

hello everybody



i have a query regarding regarding file transfer using asterisk server
, first of all i am not even sure if this is possible ....

secondly if this is possible then i want to know what kind of client
will we be using to communicate with the asterisk server in order to
transmit a file.

like for example mostly we are using the sip phones as our
clients to communicate with the server ...... kindly help me in this
regard that what kind of clients do we require for file transfer.

i dont think sip phones can be used to transfer a file ......

(maybe we dont need any client ..... but if we dont need a client then
what will we use to communicate with the server)


i will be glad if someone can guide me in this regard.


Ambreen Sheikh

Hi,

I have an issue where I don't seem to be able to dial mobile phones
from my Primary Rate service supplied by telstra.

I can dial any local numbers using 9 digits (73001xxxx) interstate I
have to dial 39415xxxx (no leading zero) and for mobiles I have tried
0417xxxxxx, 417xxxxxx, neither of which worked, just get back a
"Channel 1/1, span 1 got hangup request, cause 31".

Telstra claim that it is my equipment as the service is fully open,
so I am looking for some hints as to what I need to do to debug the
issue. I have googled around but not really found anything useful.

So I guess I might have 2 issues that might be related.
1. only having to dial 39415xxxx instead of 03 9415 xxxx
2. not being able to dial mobiles at all.

Any help greatly appreciated.
My environment is :-
Digium TE110p (the card I got on Bootcamp)
Centos 4.6
asterisk-1.4.18
zaptel-1.4.9
libpri-1.4.3

Zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
loadzone = au
defaultzone=au


Zapata.conf
;
; Zapata telephony interface
;
; Configuration file
;
; You need to restart Asterisk to re-configure the Zap channel
; CLI> reload chan_zap.so
;will reload the configuration file,
;but not all configuration options are
; re-configured during a reload.



[trunkgroups]
trunkgroup => 1,16
spanmap => 1,1,1

[channels]

context=inside

; Setup for the Primary rate card
echocancel=yes
echocancelwhenbridged=no
echotraining=yes
switchtype=EUROISDN
context=from_outside
signalling=pri_cpe
group=3
channel => 1-15
channel => 17-31
callerid=asrecieved
pridialplan=unknown
prilocaldialplan=unknown
overlapdial=yes

signalling=fxo_ls
rxwink=300; Atlas seems to use long (250ms) winks
hidecallerid=no
callwaiting=yes
restrictcid=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1

immediate=no

Relevant part of the dial plan
exten => _xxxxxxxxxx,1,noop(${EXTEN} --- ${CALLERID(num)})
exten => _xxxxxxxxxx,n,set(CALLERID(num)=73231${CALLERID(num)})
exten => _xxxxxxxxxx,n,dial(zap/g3/${EXTEN:-9})
exten => _[345]xxxxxxx,1,noop(${EXTEN} --- ${CALLERID(num)})
exten => _[345]xxxxxxx,n,set(CALLERID(num)=73231${CALLERID(num)})
exten => _[345]xxxxxxx,n,dial(zap/g3/7${EXTEN})

--- In asterisk-anz@yahoogroups.com, "morantaylor" <morantaylor@...>
wrote:
>
>
> Here is details I use with Trixbox they should be similar to what you
> put in the config files (copied out of sip_additional.conf)
>
> [039024xxxx]
> username=039024xxxx
> type=user
> secret=password
> host=sip.internode.on.net
> context=from-trunk
>
> [Nodephone]
> username=039024xxxx
> type=peer
> secret=secret
> insecure=very
> host=sip.internode.on.net
> fromuser=039024xxxx
> fromdomain=sip.internode.on.net
> dtmfmode=rfc2833
> disallow=all
> canreinvite=no
> allow=g729
> call-limit=50
>
>
> If you still have issues you could try posting on Whirlpool there are no
> doubt users who use Nodephone and Asterisk there
>
> www.whirlpool.net.au
>
> --- In asterisk-anz@yahoogroups.com, "splat000aus2000"
> <b_l_kirkness@> wrote:
> >
> > --- In asterisk-anz@yahoogroups.com, Trevor Glen tg@ wrote:
> > >
> > > Hi Brett,
> > >
> > > I'm not a Trixbox user, but I do use NodePhone via asterisk.
> > >
> > > splat000aus2000 wrote:
> > > > allow=g729
> > >
> > > Do you have the g729 codec installed in Trixbox?
> > > >
> > > > Registration
> > >
> > > If you log in to the console and type 'sip show registry' what do
> > you see?
> > >
> > > Trev
> > > --
> > > Trevor Glen
> > > Customer Relationship Manager
> > > Sarugo Pty Ltd
> > >
> > > Phone: +61 410 634 678
> > > Email: mailto:tg@
> > > Web: http://www.sarugo.net/
> > > Jabber: xmpp:tg@
> > >
> > Hi Trevor
> > I have managed to get the trunk to register. it was a default route
> > issue in linux.
> > I still cant however make an outgoing call. Can however get an
> > incomming call OK.
> > Any suggestions
> >
> > Brett
> >
>
have tried config listed above adjusted to suit my details but still
no go. I can connect using a Softphone set to call nodephone direct so
problem is with trixbox config

Brett

Here is details I use with Trixbox they should be similar to what you
put in the config files (copied out of sip_additional.conf)

[039024xxxx]
username=039024xxxx
type=user
secret=password
host=sip.internode.on.net
context=from-trunk

[Nodephone]
username=039024xxxx
type=peer
secret=secret
insecure=very
host=sip.internode.on.net
fromuser=039024xxxx
fromdomain=sip.internode.on.net
dtmfmode=rfc2833
disallow=all
canreinvite=no
allow=g729
call-limit=50


If you still have issues you could try posting on Whirlpool there are no
doubt users who use Nodephone and Asterisk there

www.whirlpool.net.au

--- In asterisk-anz@yahoogroups.com, "splat000aus2000"
<b_l_kirkness@...> wrote:
>
> --- In asterisk-anz@yahoogroups.com, Trevor Glen tg@ wrote:
> >
> > Hi Brett,
> >
> > I'm not a Trixbox user, but I do use NodePhone via asterisk.
> >
> > splat000aus2000 wrote:
> > > allow=g729
> >
> > Do you have the g729 codec installed in Trixbox?
> > >
> > > Registration
> >
> > If you log in to the console and type 'sip show registry' what do
> you see?
> >
> > Trev
> > --
> > Trevor Glen
> > Customer Relationship Manager
> > Sarugo Pty Ltd
> >
> > Phone: +61 410 634 678
> > Email: mailto:tg@
> > Web: http://www.sarugo.net/
> > Jabber: xmpp:tg@
> >
> Hi Trevor
> I have managed to get the trunk to register. it was a default route
> issue in linux.
> I still cant however make an outgoing call. Can however get an
> incomming call OK.
> Any suggestions
>
> Brett
>

splat000aus2000 wrote:
> Hi Trevor
> I have managed to get the trunk to register. it was a default route
> issue in linux.
> I still cant however make an outgoing call. Can however get an
> incomming call OK.
> Any suggestions

I think doing a "sip debug" would help.

So (again this may be different for Trixbox):

1. Log in to the asterisk console
2. Type sip debug
3. Try to make the call.
4. Capture the output.

Something like that.

HTH,
Trev
--
Trevor Glen
Customer Relationship Manager
Sarugo Pty Ltd

Phone:  +61 410 634 678
Email:  mailto:tg@...
Web:    http://www.sarugo.net/
Jabber: xmpp:tg@...

--- In asterisk-anz@yahoogroups.com, Trevor Glen <tg@...> wrote:
>
> Hi Brett,
>
> I'm not a Trixbox user, but I do use NodePhone via asterisk.
>
> splat000aus2000 wrote:
> > allow=g729
>
> Do you have the g729 codec installed in Trixbox?
> >
> > Registration
>
> If you log in to the console and type 'sip show registry' what do
you see?
>
> Trev
> --
> Trevor Glen
> Customer Relationship Manager
> Sarugo Pty Ltd
>
> Phone:  +61 410 634 678
> Email:  mailto:tg@...
> Web:    http://www.sarugo.net/
> Jabber: xmpp:tg@...
>
Hi Trevor
I have managed to get the trunk to register. it was a default route
issue in linux.
I still cant however make an outgoing call. Can however get an
incomming call OK.
Any suggestions

Brett

Hi Brett,

I'm not a Trixbox user, but I do use NodePhone via asterisk.

splat000aus2000 wrote:
> allow=g729

Do you have the g729 codec installed in Trixbox?
>
> Registration

If you log in to the console and type 'sip show registry' what do you see?

Trev
--
Trevor Glen
Customer Relationship Manager
Sarugo Pty Ltd

Phone:  +61 410 634 678
Email:  mailto:tg@...
Web:    http://www.sarugo.net/
Jabber: xmpp:tg@...

--- In asterisk-anz@yahoogroups.com, "thegreenembrace" <daniel@...> wrote:
>
> i keep asking questions and they just disappear, why does this happen,
> i thought that they may not be Australian enough but others ask non
> specific Australian questions. why are mine invalid, i don't even get
> a a sorry your question has been rejected because....
>
> very unhappy and now also sad.
> daniel
>

I just checked and your listed as unmoderated so your messages should
just come through.

--- In asterisk-anz@yahoogroups.com, "splat000aus2000"
<b_l_kirkness@...> wrote:
>
> I have posted 2 on topic questions to the group but they have both
> just disappeared. What is happening?
>

Not deleted.

Just held in a moderation queue.

You should be good to go now.

Hi all
I know this may be a simple problem to fix but I am having trouble
getting my Trixbox to connect to Nodephone.
I can get a softphone to connect and make and recieve calls so I know
that there is no problem at the VSP but cannot get a registration via
Trixbox.

Here are the trunk settings.


Trunk name = Nodephone

allow=g729
canreinvite=no
disallow=all
dtmfmode=rfc2833
fromdomain=sip.internode.on.net
fromuser=02904XXXXX
host=sip.internode.on.net
insecure=very
secret=XXXXXXXXX
type=peer

User Context = nodephone number

context=from-trunk
host=sip.internode.on.net
secret=XXXXXXXXX
type=user
username=02904XXXXX



Registration

02904XXXXX:XXXXXXXXX@...


any help getting this to work would be appreciated.

Regards Brett Kirkness





[Non-text portions of this message have been removed]

I have asterisk setup with a single trunk that is registered with my
VSP but I am unable to make any outgoing call. I get a message saying
no available trunks.

Incoming calls work so I suspect there is a problem with my outbound
calling setup and not with the trunk itself.

I have posted 2 on topic questions to the group but they have both
just disappeared. What is happening?

Thanks Mike the problem has been solved.

--- In asterisk-anz@yahoogroups.com, "Mike" <mike@...> wrote:
>
> If you really spell "allow" as "alow",  then you may not be enabling
any codecs for outgoing calls.  That could have an effect.
>
>
>
>
>   ----- Original Message -----
>   From: don juan de marco
>   To: asterisk-anz@yahoogroups.com
>   Sent: Wednesday, October 31, 2007 3:41 PM
>   Subject: [asterisk-anz] ASTERISK+WDP+ SIP TRUNK settings
>
>
>   Hi all
>
>   We have been experiencing problems with our Asterisk and voip
>   provider. The problem is we can not make it work at all. The version
>   we are running is 1.2.13. The dial tone could be heard on the voip
>   handset but on the actual phone that is receiving the call no tone can
>   be heard. These are the details from our SIP Trunk and sip.conf;
>
>   SIP TRUNK SETTINGS
>
>   DIAL RULES
>
>   601XXXXXXXX
>   60NXXXXXXX
>   61+4XXXXXXXX
>   899060X.
>   60ZXX.
>   06612+NXXXXXXX
>   0661+NXXXXXXXX
>   61+1300XXXXXX
>   61+13ZXXX
>   61+1800XXXXXX
>
>   OUTGOING SETTINGS
>
>   Peer Details
>
>   alow=g729&ulaw&alaw
>   authuser=6199xxxx
>   context=from-trunk
>   disallow=all
>   fromdomain=sipau2.worlddialpoint.net
>   fromuser=6199xxxx
>   host=202.168.56.133
>   insecure=very
>   qualify=yes
>   secret=xxxx
>   type=peer
>   username=6199xxxx
>
>   INCOMING SETTINGS
>
>   canreinvite=no
>   context=from-trunk
>   fromuser=6199xxxx
>   insecure=very
>   qualify=yes
>   secret=xxxx
>   type=user
>   username=6199xxxx
>
>   REGISTRATION
>
>   6199xxx:xxx@.../6199xxx
>
>   SIP CONFIG
>
>   ; Note: If your SIP devices are behind a NAT and your Asterisk
>   ; server isn't, try adding "nat=1" to each peer definition to
>   ; solve translation problems.
>
>   [general]
>   port=5060 ; Port to bind to (SIP is 5060)
>   bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine)
>   nat=yes
>   disallow=all
>   alow=g729
>   allow=ulaw
>   allow=alaw
>   insecure=very
>   useragent=anything
>   maxexpirey=3600
>   defaultexpirey=600
>   context = from-trunk ; Send unknown SIP callers to this context
>   callerid = Unknown
>   tos=0x68
>
>   ; If you need to answer unauthenticated calls, you should change this
>   ; next line to 'from-trunk', rather than 'from-sip-external'.
>   ; You'll know this is happening if when you call in you get a message
>   ; saying "The number you have dialed is not in service. Please
check the
>   ; number and try again."
>
>   ; #, in this configuration file, is NOT A COMMENT. This is exactly
>   ; how it should be.
>
>   #include sip_nat.conf
>   #include sip_custom.conf
>   #include sip_additional.conf
>
>   If anyone could help us get this baby going we would appreciate it
>   very much. If further details are required for you to diagnose it
>   please tell us what details are required.
>
>   Thank you.
>
>
>
>
>
> [Non-text portions of this message have been removed]
>

If you really spell "allow" as "alow",  then you may not be enabling any codecs
for outgoing calls.  That could have an effect.




   ----- Original Message -----
   From: don juan de marco
   To: asterisk-anz@yahoogroups.com
   Sent: Wednesday, October 31, 2007 3:41 PM
   Subject: [asterisk-anz] ASTERISK+WDP+ SIP TRUNK settings


   Hi all

   We have been experiencing problems with our Asterisk and voip
   provider. The problem is we can not make it work at all. The version
   we are running is 1.2.13. The dial tone could be heard on the voip
   handset but on the actual phone that is receiving the call no tone can
   be heard. These are the details from our SIP Trunk and sip.conf;

   SIP TRUNK SETTINGS

   DIAL RULES

   601XXXXXXXX
   60NXXXXXXX
   61+4XXXXXXXX
   899060X.
   60ZXX.
   06612+NXXXXXXX
   0661+NXXXXXXXX
   61+1300XXXXXX
   61+13ZXXX
   61+1800XXXXXX

   OUTGOING SETTINGS

   Peer Details

   alow=g729&ulaw&alaw
   authuser=6199xxxx
   context=from-trunk
   disallow=all
   fromdomain=sipau2.worlddialpoint.net
   fromuser=6199xxxx
   host=202.168.56.133
   insecure=very
   qualify=yes
   secret=xxxx
   type=peer
   username=6199xxxx

   INCOMING SETTINGS

   canreinvite=no
   context=from-trunk
   fromuser=6199xxxx
   insecure=very
   qualify=yes
   secret=xxxx
   type=user
   username=6199xxxx

   REGISTRATION

   6199xxx:xxx@.../6199xxx

   SIP CONFIG

   ; Note: If your SIP devices are behind a NAT and your Asterisk
   ; server isn't, try adding "nat=1" to each peer definition to
   ; solve translation problems.

   [general]
   port=5060 ; Port to bind to (SIP is 5060)
   bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine)
   nat=yes
   disallow=all
   alow=g729
   allow=ulaw
   allow=alaw
   insecure=very
   useragent=anything
   maxexpirey=3600
   defaultexpirey=600
   context = from-trunk ; Send unknown SIP callers to this context
   callerid = Unknown
   tos=0x68

   ; If you need to answer unauthenticated calls, you should change this
   ; next line to 'from-trunk', rather than 'from-sip-external'.
   ; You'll know this is happening if when you call in you get a message
   ; saying "The number you have dialed is not in service. Please check the
   ; number and try again."

   ; #, in this configuration file, is NOT A COMMENT. This is exactly
   ; how it should be.

   #include sip_nat.conf
   #include sip_custom.conf
   #include sip_additional.conf

   If anyone could help us get this baby going we would appreciate it
   very much. If further details are required for you to diagnose it
   please tell us what details are required.

   Thank you.





[Non-text portions of this message have been removed]

The following works fine for me.  I use WDP for outgoing landline calls from
Sydney to Australian national numbers.




Dial Rules:

0+NXXXXXXXX



Outgoing:

allow=alaw
authuser=6199xxxxxxx
context=from-trunk
disallow=all
dtmfmode=inband
fromdomain=sip.worlddialpoint.net
fromuser=6199xxxxxxx
host=202.168.56.133
insecure=very
secret=password
type=peer
username=6199xxxxxxx


Incoming


canreinvite=no
context=from-trunk
fromuser=6199xxxxxxx
insecure=very
qualify=no
secret=password
type=user
username=6199xxxxxxx



Reg:

6199xxxxxxx:password@.../6199xxxxxxx


SIP.CONF

; Note: If your SIP devices are behind a NAT and your Asterisk
;  server isn't, try adding "nat=1" to each peer definition to
;  solve translation problems.

[general]

port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=alaw
allow=ulaw
allow=g723
allow=g729
allow=ilbc
allow=gsm
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-pstn ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68
useragent=MyPABX

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf



RTP.CONF

;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=10001
rtpend=10100

SIP.NAT.CONF

nat=yes
externhost=xxxxxxxx.homelinux.net
localnet=192.168.1.0/255.255.255.0
externrefresh=4



   ----- Original Message -----
   From: don juan de marco
   To: asterisk-anz@yahoogroups.com
   Sent: Wednesday, October 31, 2007 3:41 PM
   Subject: [asterisk-anz] ASTERISK+WDP+ SIP TRUNK settings


   Hi all

   We have been experiencing problems with our Asterisk and voip
   provider. The problem is we can not make it work at all. The version
   we are running is 1.2.13. The dial tone could be heard on the voip
   handset but on the actual phone that is receiving the call no tone can
   be heard. These are the details from our SIP Trunk and sip.conf;

   SIP TRUNK SETTINGS

   DIAL RULES

   601XXXXXXXX
   60NXXXXXXX
   61+4XXXXXXXX
   899060X.
   60ZXX.
   06612+NXXXXXXX
   0661+NXXXXXXXX
   61+1300XXXXXX
   61+13ZXXX
   61+1800XXXXXX

   OUTGOING SETTINGS

   Peer Details

   alow=g729&ulaw&alaw
   authuser=6199xxxx
   context=from-trunk
   disallow=all
   fromdomain=sipau2.worlddialpoint.net
   fromuser=6199xxxx
   host=202.168.56.133
   insecure=very
   qualify=yes
   secret=xxxx
   type=peer
   username=6199xxxx

   INCOMING SETTINGS

   canreinvite=no
   context=from-trunk
   fromuser=6199xxxx
   insecure=very
   qualify=yes
   secret=xxxx
   type=user
   username=6199xxxx

   REGISTRATION

   6199xxx:xxx@.../6199xxx

   SIP CONFIG

   ; Note: If your SIP devices are behind a NAT and your Asterisk
   ; server isn't, try adding "nat=1" to each peer definition to
   ; solve translation problems.

   [general]
   port=5060 ; Port to bind to (SIP is 5060)
   bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine)
   nat=yes
   disallow=all
   alow=g729
   allow=ulaw
   allow=alaw
   insecure=very
   useragent=anything
   maxexpirey=3600
   defaultexpirey=600
   context = from-trunk ; Send unknown SIP callers to this context
   callerid = Unknown
   tos=0x68

   ; If you need to answer unauthenticated calls, you should change this
   ; next line to 'from-trunk', rather than 'from-sip-external'.
   ; You'll know this is happening if when you call in you get a message
   ; saying "The number you have dialed is not in service. Please check the
   ; number and try again."

   ; #, in this configuration file, is NOT A COMMENT. This is exactly
   ; how it should be.

   #include sip_nat.conf
   #include sip_custom.conf
   #include sip_additional.conf

   If anyone could help us get this baby going we would appreciate it
   very much. If further details are required for you to diagnose it
   please tell us what details are required.

   Thank you.





[Non-text portions of this message have been removed]

Hi all

We have been experiencing problems with our Asterisk and voip
provider. The problem is we can not make it work at all. The version
we are running is 1.2.13. The dial tone could be heard on the voip
handset but on the actual phone that is receiving the call no tone can
be heard. These are the details from our SIP Trunk and sip.conf;

SIP TRUNK SETTINGS

DIAL RULES

601XXXXXXXX
60NXXXXXXX
61+4XXXXXXXX
899060X.
60ZXX.
06612+NXXXXXXX
0661+NXXXXXXXX
61+1300XXXXXX
61+13ZXXX
61+1800XXXXXX

OUTGOING SETTINGS

Peer Details

alow=g729&ulaw&alaw
authuser=6199xxxx
context=from-trunk
disallow=all
fromdomain=sipau2.worlddialpoint.net
fromuser=6199xxxx
host=202.168.56.133
insecure=very
qualify=yes
secret=xxxx
type=peer
username=6199xxxx

INCOMING SETTINGS

canreinvite=no
context=from-trunk
fromuser=6199xxxx
insecure=very
qualify=yes
secret=xxxx
type=user
username=6199xxxx

REGISTRATION

6199xxx:xxx@.../6199xxx


SIP CONFIG

; Note: If your SIP devices are behind a NAT and your Asterisk
;  server isn't, try adding "nat=1" to each peer definition to
;  solve translation problems.

[general]
port=5060 	 ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0  ; Address to bind to (all addresses on machine)
nat=yes
disallow=all
alow=g729
allow=ulaw
allow=alaw
insecure=very
useragent=anything
maxexpirey=3600
defaultexpirey=600
context = from-trunk ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."



; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf


If anyone could help us get this baby going we would appreciate it
very much. If further details are required for you to diagnose it
please tell us what details are required.

Thank you.

Hi

Have you tried Send DTMF via SIP INFO message in your ATA configuration?

Regards,

Halomoan

On 9/22/07, James <james@...> wrote:
> I seem to remember that the hookflash transfer dose work with the ZyXel
> P2002. I will ask one of my users that uses one next week for you. Is that
> the silver model or the black one. You should also look at firmware upgrades
> for the ZyXel as they have come a long way since the first ones.
> I also seem to remember that the fix for the dtmf tones was in a Asterisk
> patch at some stage. I wasn't the tech that did that job but I think that
> what he said did it. Anyway its a place for you to start. We defiantly have
> DTMF tones working on our system so someone in the office will know the
> answer on Monday.
> Good luck
>
> ----- Original Message -----
> From: "Richard" <rich_lists@...>
> To: <asterisk-anz@yahoogroups.com>
> Sent: Saturday, September 22, 2007 3:16 PM
> Subject: [asterisk-anz] Asterisk issues in very simple configuration.
>
>
> >I have an italk trunk, and a softphone, and an ata.
> >
> >
> >
> > Its set up so that incoming calls will ring both the softphone and the
> > ata,
> > after 20 seconds of no answer it does an answer, plays something to say im
> > not there to answer and to try again later and hangs up (I hate listening
> > to
> > voicemails)
> >
> >
> >
> > Anyway.
> >
> >
> >
> > If a call comes in and I take it on the phone on the ATA, I cant send
> > DTMF's
> > out, and if I press # (Which I have to do to acknowledge some incoming
> > calls
> > as being received) then asterisk just says TRANSFER and puts the incoming
> > call on hold. And gives me a dialtone. Any other button just gives
> > clicking
> > noises to the outside party.
> >
> >
> >
> > Now to me, a transfer is done with a press of the flash button, not #, and
> > I
> > don't really need this functionality at the moment. Most critical is that
> > I
> > can sent DTMF to incoming calls. The softphone (X-pro) can do so.
> >
> >
> >
> > The ata is a piece of junk zyxel p2002 - but dtmf works ok when its
> > directly
> > registered to italk. I just want to have it go thru asterisk. I don't care
> > if I lose transfer ability, since I don't think this ATA supports it at
> > all.
> >
> >
> >
> > All dtmfs are set to RFC in the sip.conf. Not sure what else to change to
> > get this working properly.
> >
> >
> >
> > If I need a specific type of ATA to get a normal  working
> > in asterisk then please tell me one that's good since I will be buying
> > some
> > more soon for other parts of the house.
> >
> >
> >
> > Thanks.
> >
> >
> >
> > [Non-text portions of this message have been removed]
> >
> >
> >
> >
> > Yahoo! Groups Links
> >
> >
> >
> >
> >
> > --
> > No virus found in this incoming message.
> > Checked by AVG Free Edition.
> > Version: 7.5.487 / Virus Database: 269.13.27/1020 - Release Date:
> > 9/20/2007 12:07 PM
> >
> >
>
>
>
>
> Yahoo! Groups Links
>
>
>
>

I seem to remember that the hookflash transfer dose work with the ZyXel
P2002. I will ask one of my users that uses one next week for you. Is that
the silver model or the black one. You should also look at firmware upgrades
for the ZyXel as they have come a long way since the first ones.
I also seem to remember that the fix for the dtmf tones was in a Asterisk
patch at some stage. I wasn't the tech that did that job but I think that
what he said did it. Anyway its a place for you to start. We defiantly have
DTMF tones working on our system so someone in the office will know the
answer on Monday.
Good luck

----- Original Message -----
From: "Richard" <rich_lists@...>
To: <asterisk-anz@yahoogroups.com>
Sent: Saturday, September 22, 2007 3:16 PM
Subject: [asterisk-anz] Asterisk issues in very simple configuration.


>I have an italk trunk, and a softphone, and an ata.
>
>
>
> Its set up so that incoming calls will ring both the softphone and the
> ata,
> after 20 seconds of no answer it does an answer, plays something to say im
> not there to answer and to try again later and hangs up (I hate listening
> to
> voicemails)
>
>
>
> Anyway.
>
>
>
> If a call comes in and I take it on the phone on the ATA, I cant send
> DTMF's
> out, and if I press # (Which I have to do to acknowledge some incoming
> calls
> as being received) then asterisk just says TRANSFER and puts the incoming
> call on hold. And gives me a dialtone. Any other button just gives
> clicking
> noises to the outside party.
>
>
>
> Now to me, a transfer is done with a press of the flash button, not #, and
> I
> don't really need this functionality at the moment. Most critical is that
> I
> can sent DTMF to incoming calls. The softphone (X-pro) can do so.
>
>
>
> The ata is a piece of junk zyxel p2002 - but dtmf works ok when its
> directly
> registered to italk. I just want to have it go thru asterisk. I don't care
> if I lose transfer ability, since I don't think this ATA supports it at
> all.
>
>
>
> All dtmfs are set to RFC in the sip.conf. Not sure what else to change to
> get this working properly.
>
>
>
> If I need a specific type of ATA to get a normal  working
> in asterisk then please tell me one that's good since I will be buying
> some
> more soon for other parts of the house.
>
>
>
> Thanks.
>
>
>
> [Non-text portions of this message have been removed]
>
>
>
>
> Yahoo! Groups Links
>
>
>
>
>
> --
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.5.487 / Virus Database: 269.13.27/1020 - Release Date:
> 9/20/2007 12:07 PM
>
>

I have an italk trunk, and a softphone, and an ata.



Its set up so that incoming calls will ring both the softphone and the ata,
after 20 seconds of no answer it does an answer, plays something to say im
not there to answer and to try again later and hangs up (I hate listening to
voicemails)



Anyway.



If a call comes in and I take it on the phone on the ATA, I cant send DTMF's
out, and if I press # (Which I have to do to acknowledge some incoming calls
as being received) then asterisk just says TRANSFER and puts the incoming
call on hold. And gives me a dialtone. Any other button just gives clicking
noises to the outside party.



Now to me, a transfer is done with a press of the flash button, not #, and I
don't really need this functionality at the moment. Most critical is that I
can sent DTMF to incoming calls. The softphone (X-pro) can do so.



The ata is a piece of junk zyxel p2002 - but dtmf works ok when its directly
registered to italk. I just want to have it go thru asterisk. I don't care
if I lose transfer ability, since I don't think this ATA supports it at all.



All dtmfs are set to RFC in the sip.conf. Not sure what else to change to
get this working properly.



If I need a specific type of ATA to get a normal hookflash transfer working
in asterisk then please tell me one that's good since I will be buying some
more soon for other parts of the house.



Thanks.



[Non-text portions of this message have been removed]

I am using g729 with 4 PennyTel accounts to call international and its working
fine.

   about your second note, i am note sure but try to add the same trunk twice but
with different codec each time, remember not to put the register string as you
can't regiser twice for the same account.

   Regards,
   Ammar


Matt Rees <matt@...> wrote:
           Hello everyone
First time poster, long time reader :)

Just wondering if anyone has any advice on best codecs to use for
International Phone Calls.

I used to use G711U (Pennytel) for all calls (national and
international). But I seem to be running into a few problems with them
lately on the international front.

If anyone has any advice on best international codecs (we mainly call
US and Canada). Id really appreciate hearing it.

On a 2nd note, if anyone knows the best way to use different codecs for
different routes (not trunks), Id like to hear any advice from you also.

Thanks for your time..

Hoping all is well,

Matt






---------------------------------
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[Non-text portions of this message have been removed]

i keep asking questions and they just disappear, why does this happen,
i thought that they may not be Australian enough but others ask non
specific Australian questions. why are mine invalid, i don't even get
a a sorry your question has been rejected because....

very unhappy and now also sad.
daniel

Hello everyone
First time poster, long time reader :)

Just wondering if anyone has any advice on best codecs to use for
International Phone Calls.

I used to use G711U (Pennytel) for all calls (national and
international). But I seem to be running into a few problems with them
lately on the international front.

If anyone has any advice on best international codecs (we mainly call
US and Canada). Id really appreciate hearing it.

On a 2nd note, if anyone knows the best way to use different codecs for
different routes (not trunks), Id like to hear any advice from you also.

Thanks for your time..

Hoping all is well,

Matt

Is there anyway to record calls by extension?
I do not need to use a dial plan or any other static configuration.

I am using Asterix.I need to use phone extension instead of channel
name of the phone for monitor command.Basically,I need to record voice
converstaions using extensions.In Asterix,I see ,it can be done using
monitor command.But,it needs channel name.It is a bit long process to
capture channel name and extract extenstion name.

Is there any easy way to record voice converstaions by phone extension?

Hi,
does anybody have an well working Cisco 7970 together with Asterisk?
I am looking for someone who is able to create the config-file.
The Asterisk itself will be outside of the firewall and has different
accounts.
The phone is changed from Skinny to SIP with the last OS-version.

Please let me know if you can help (it's only one phone).
Also it would be of interest to use a phone book together with one of
the  typical Asterisk admin-tools (e.g. AsteriskNow GUI or probably
better: FreePBX).
Can you recommend a solution, can you implement it?

Maybe you can install Asterisk (1.4.x) completely (because I want to
install it completely new on a Centos5 platform where already some
other applications working).

The most important part is the Cisco phone (currently I use the
Grandstream)...

Best regards
Jochen

Hi,

I'm sorry, just got the answer.
1. Create an extension, e.g.: 222
2. Call *72, set forwarding for extension 222 to teleconference room 8222.
3. Any call from PSTN to 222 will be automatically redirected to conference room
8222.

Thanks,
Hendra


----- Original Message ----
From: "cyhss@..." <cyhss@...>
To: asterisk-anz@yahoogroups.com
Sent: Tuesday, August 7, 2007 9:47:14 PM
Subject: [asterisk-anz] Teleconference using Asterisk @ Home













             Hi,



I try to make a teleconference room using Asterisk @ Home, however the inbound
call always fail to reach 8xxx number. Does anyone know how to reach the
teleconference room from PSTN line?

Any help is appreciated.



Thanks,

Hendra Santosa



____________ _________ _________ _________ _________ _________ _

Park yourself in front of a world of choices in alternative vehicles. Visit the
Yahoo! Auto Green Center.

http://autos. yahoo.com/ green_center/



[Non-text portions of this message have been removed]














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[Non-text portions of this message have been removed]

Hi,

I try to make a teleconference room using Asterisk @ Home, however the inbound
call always fail to reach 8xxx number. Does anyone know how to reach the
teleconference room from PSTN line?
Any help is appreciated.

Thanks,
Hendra Santosa




      
________________________________________________________________________________\
____
Park yourself in front of a world of choices in alternative vehicles. Visit the
Yahoo! Auto Green Center.
http://autos.yahoo.com/green_center/

[Non-text portions of this message have been removed]

--- In asterisk-anz@yahoogroups.com, "mobileshea" <sheasie@...>
wrote:
>
hi its me here, i have 1 year working expereince in asterisk
communication with PRI and SS7 and call center development solution.
as well as hardware plateform selection. os selection and other
asterisk related hardware selection expereince. if u still need the
eprson thn i m here, i can remotly installa nd make prepare the
asterisk ssytem with all modules, so it wd be ready for
communication witha ll configuration.


regards,
jehanzaib younis
+92.111.874.874 Ext: 2558
+92.321.5139853
jehanzaib.younis@...
http://www.trgcustomersolutions.com


i m here available ...
> Dear Group Members,
>
> Eventually, I will be looking for someone to help install,
configure
> and customize an Asterisk-based system.  I am curious to know what
to
> expect in terms of fees/rates.
>
> But in the mean time, I am hoping someone could offer me just 30
> minutes of free/evaluation time, as I am totally new to this
> environment, and I do not even know if Asterisk is capable of doing
> what it is that I require.
>
> Looking forward to any thoughts, suggestions,
>
> Respectfully,
>
> Shea
>