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Problem with SendDTMF

Hello,

I am having a problem with SendDTMF – it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.

I use software phone to test it… when I dialed 501, I cant hear anything
for about 10 seconds (this is because of SendDTMF) and then I can hear
the operator saying to enter the numbers but SendDTMF already did it?!

Asterisk server are connected to voip.ms provider.

I have spent many hours trying to get to work, how to fix this issue?

See the configuration and debug log below:

extensions.conf
================
[test]
exten => 501,1,Set(CALLERID(num)=004471XXXXXXX)
exten => 501,n,Dial(SIP/+44797XXXXXX@voipms,30,M(sendnumber)t)
exten => 501,n,Hangup()

[macro-sendnumber]
exten => s,1,Wait(3)
exten => s,n,SendDTMF(www0w7w8w8wXwXwXwXwXwX)

sip.conf
==========
[general]
context=default
tcpbindaddr=0.0.0.0
dtmfmode = rfc2833
register => xxxxx:vxxxxx@london.voip.ms:5060

[test]
type=peer
secret=2xxx
host=dynamic
context=test

[voipms]
canreinvite=no
host=london.voip.ms
secret=xxxxxx
type=peer
username=135xxx ;your account
disallow=all
allow=gsm
; allow=g729 ; Uncomment if you support G729
fromuser=135xxx
insecure=invite
trustrpid=yes
sendrpid=yes
nat=yes
dtmfmode=rfc2833

debug:
=====
== Using SIP RTP CoS mark 5

Grandstream 1.0.3.30 BETA Firmware

If you are using Grandstream GXP 21XXX and 14XX phones and you are doing
any kind of remote firmware updates or config updates DO NOT use the
1.0.3.30 BETA version. We have found a bug in it that causes HTTP updates
to not work if you are using a domain name and not an IP address for
pulling configs and firmware. It can put you in a state where you can’t
update the configs or firmware without direct web or telnet contact to the
phone.

Thanks

Bryant

Asterisk ACL

Hi

We are trying to accept inbound calls from a SIP provider who sends us calls from various IP’s within a given subnet but they are failing every time with the following message on the console.

chan_sip.c:20006 handle_request_invite: Call from ” to extension ‘‘ rejected because extension not found

Our understanding is that the deny line blocks every IP and the following permit line then allows calls from the specified subnet but it seems that the peer is never matched when a calls hits the server.
It’s almost as if there should be a setting somewhere that we are missing to enable ACL’s.

Can anyone point us in the right direction here please? Is our understanding simply not correct?

In our peer config we have:

host = dynamic
type = peer
deny = 0.0.0.0/0.0.0.0
permit = xxx.xxx.xxx.xxx/255.255.255.0
context = Test
insecure = invite,port

Thanks in advance
Mark.

in sip.conf or dialplan or db?

On Tuesday 27 March 2012, Roland wrote:
> I am setting up my dialplan with quite some outbound numbers. We have a
> block of 100 DID’s, for which some of them will go direct to specific
> phones. I am struggling how to solve this, so I am searching for a little
> advice. These are my concerns.
>
> I could set the DID in the sip.conf using something like:
>
> callerid=”137-Roland” <31229253137>
>
> 137 would be my extention number here.
> …..
> Any suggestions would be appreciated! Am I missing any options here?

Assuming there is a reasonable mapping from “outside” numbers to “inside”
numbers (your example shows the external number as being 31229253 followed by
the internal number; note also you can do simple integer maths within a
dialplan, as unlike some languages + hasn’t been appropriated for string
concatenation), it shouldn’t be that difficult.

If an internal extension wants to call an internal extension, its caller ID
ought already to be set to its “inside” extension number in sip.conf. If it
wants to call an external number, then it needs to set its caller ID to the
appropriate inbound number. If it’s as simple as adding a prefix, then you
need something in your dialplan like:

[management]
; 3 digits is internal
exten => XXX,1,Dial(SIP/${EXTEN},90)
exten => XXX,2,Hangup()
; 5 digits or more must be external — ident as own DDI number
exten => XXXX.,1,Set(CallerID(num)=${DDIPREFIX}${callerID(num)})
exten => XXXX.,2,Dial(${POTS}/${EXTEN},90)
exten => XXXX.,3,Hangup()

Now, chances are, not every internal phone will want to ident as a unique
external number — perhaps you want all the phones in one office to ident as a
single number, and all ring together (using something like
Dial(TECH/ext&TECH/ext&TECH/ext ….. ) when called from outside. In this
case, you just need to configure all these phones into their own context in
your sip.conf and something like this in your dialplan:

[sales-office]
; 3 digits is internal
exten => XXX,1,Dial(SIP/${EXTEN},90)
exten => XXX,2,Hangup()
; 5 digits or more must be external — one ident for whole office
exten => XXXX.,1,Set(CallerID(num)=${SALESOFFICE})
exten => XXXX.,2,Dial(${POTS}/${EXTEN},90)
exten => XXXX.,3,Hangup()

Park and PARKINGDYNAMIC

I have been trying to get the dynamic parking working.

For some reason when I park a call using this method the console says it is
using the default parking context not the one I am trying to specidfy. It
also is playing the parked extension to the caller. I am transfering the
call to an extension that is doing a goto to the context below. Any ideas
or examples on how to make this work. What I need to be able to do is have
multiple parking lots using the same extension pools but seperated by a
dynamic context of ${account}-Lot. So that each office suite cant cross
pickup another groups parked calls while using the same number pool of
110-120. I need the dynamic option as all of our calls are database driven
and we can’t add a seperate entry per customer to the feautres.conf.

[MSIP-DynPark]
exten => s,1,NoOp(Dynamic Parking)
exten => s,n,NoOp(Return Parked Call)
exten => s,n,GoTo(${CUT(${l_ndeContext}-ndeArgs,~,1)},1)

exten => _XXX,1,Set(PARKINGDYNAMIC=parkinglot_small)
exten => _XXX,n,Set(PARKINGDYNEXTEN=110)
exten => _XXX,n,Set(PARKINGDYNPOS=111-120)
exten => _XXX,n,Set(PARKINGDYNCONTEXT=${account}-Lot)
;exten => _XXX,n,Set(PARKINGEXTEN=99)
exten => _XXX,n,Park()

[MSIP-DynParkPickup]
exten => _NXX,1,ParkedCall(${EXTEN},${account}-Lot)
exten => _NXX,hint,park:$EXTEN@${account}-Lot

Thanks

Bryant

Sporadic one way audio problem

Hi all again,

I’ve got a problem with sporadic one way audio calls, which means
sometimes I can’t hear the calling party (call is established, but audio
is missing). Today I received ~90 calls, one of them got this problem.

I’ve got two networks involved, without NAT:

- 192.168.1.X, in there one nic of my server and all the phones
- a private net to my provider, in there a nic of my server and the voice
switch of my provider

My sip.conf:
[general]
port=5060
bindaddr=0.0.0.0
language=de
allowguest=no
;echocancel=yes
;echotraining=yes
alwaysauthreject=yes
disallow=all
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=XXX.XXX.X.X/29
permit=192.168.1.0/24
;jbenable=yes
;jbforce=yes
;jbmaxsize=20
;jbresyncthreshold=1000
tos=0×10
directmedia=no
nat=no
directrtpsetup=no

[provider]
type=peer
host=XXX.XXX.X.X
insecure=port,invite
context=XXXXXXXXX
dtmfmode=rfc2833
directmedia=no
nat=no
directrtpsetup=no
;qualify=300

[one-phone]
[10]
type=peer
context=XXXXXXXXX
secret=XXXXXX
host=dynamic
;qualify=300
directmedia=no
nat=no
directrtpsetup=no
dtmfmode=inband

Any help greatly appreciated!

Thanks,
Georg