* You are viewing the archive for November 17th, 2011

2 same sip extension number on 2 asterisk – call not passing on certain condition

Hi list,

something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both
having an extension [115], one as type peer (caller side 1.4) and one as
friend (callee side 1.8). Phones from both location connect to Asterisk
from LAN. Router are Linux boxes.

Connection between the 2 sites is done like this:

On the callee side

[115] ;callee
type=friend
host=dynamic
secret=otherSecret
context=local
nat=no
canreinvite=no
qualify=no
dtmfmode=rfc2833
allow=all
call-limit=1
busy-level=1
allow=all

[Caller]
type=peer
host=voip1.domain.net
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.xxx
context=myAccess
disallow=all
allow=all
nat=yes
insecure=port,invite

On the caller side

[115]
type=peer
username=115
secret=blabla
context=local
host=dynamic
nat=yes
canreinvite=no
dtmfmode=auto
disallow=all
allow=jpeg,png,h263,h263p,h264,alaw,ulaw
callgroup=1
pickupgroup=1
insecure=invite

[Callee]
type=peer
host=voip1.other-domain.net
deny=0.0.0.0/0.0.0.0
permit=yyy.yyy.yyy.yyy
context=myOtherAccess
disallow=all
allow=all

Now when I call from 115@caller to any number at callee side I’m
rejected with

Sending to xxx.xxx.xxx.xxx:5060 (no NAT)
Using INVITE request as basis request -
281799ed7524c46966bcf303371edba4@xxx.xxx.xxx.xxx
Found peer ‘115’ for ‘115’ from xxx.xxx.xxx.xxx:5060

< --- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060 --->
SIP/2.0 401 Unauthorized

This is, Asterisk try to authenticate on URI SIP user before from peer
definition. If I change type from friend to peer it worked (I need the
friend for this extension)

Does someone has an idea on how to solve this problem?

Thanks for any hint

Question about Read() application

Hello again list,

Did the following: (on 1.4.42 installation)

asterisk -rx “core show application read”

-= Info about application ‘Read’ =-

[Synopsis]

Read a variable

[Description]

Read(variable[|filename][|maxdigits][|option][|attempts][|timeout])

Reads a #-terminated string of digits a certain number of times from the

user in to the given variable.

filename — file to play before reading digits or tone with option i

maxdigits — maximum acceptable number of digits. Stops reading after

maxdigits have been entered (without requiring the user to

press the ‘#’ key).

Defaults to 0 – no limit – wait for the user press the ‘#’
key.

Any value below 0 means the same. Max accepted value is 255.

option — options are ‘s’ , ‘i’, ‘n’

‘s’ to return immediately if the line is not up,

‘i’ to play filename as an indication tone from your
indications.conf

‘n’ to read digits even if the line is not up.

attempts — if greater than 1, that many attempts will be made in the

event no data is entered.

timeout — An integer number of seconds to wait for a digit response.
If greater

than 0, that value will override the default timeout.

Read should disconnect if the function fails or errors out.

I need for Asterisk to “keep going” if I have a problem with Read() or at
least playback some kind of warning. Any suggestions?

P.S. The “Read should disconnect” line does not show up in my 10.0
install.

Thanks

Danny Nicholas

DTMF dropping in Read Command

Hello listers,

I have a couple of 1.4.37 through 1.4.42 boxes
running at different sites. These systems run a fairly simple IVR that uses
waitexten and Read to get credentials and plow on through a set of contexts.
I am experiencing two problems in my setup:

1. In environments where DAHDI is the trunk of choice, this snippet
drops digits, especially if the user attempts to key them quickly. It also
mysteriously hangs up (autofallthrough=yes) if they key them too slowly

exten => s,1,Set(chktime=0)

exten => s,n,Read(digitacct,entacct,17,skip,1,10)

exten => s,n,Gotoif($[${LEN(${digitacct})} < 1]?checking,t,1)

exten => s,n,Read(digitpass,entpin,5,skip,1,10)

2. This context will randomly hang up the call when the user presses 1

[checking]

Exten => s,1,noop()

exten => s,n,Set(BALPLAY=${balance})

exten => s,n(chkdetail),Background(${BALPLAY})

exten => s,n(aftbal),Background(aftbal)

exten => s,n,WaitExten(6)

exten => s,n,Goto(checking,t,9)

exten => s,n,Goto(checking,s,chkdetail)

exten => 1,1,Goto(checking,s,chkdetail)

Any ideas?

Thanks in advance

Danny Nicholas

how to debug dropped calls?

I’ve been experiencing a number of dropped calls – both where I’m
calling out and the call drops before answer, and where it’s inbound and
the call drops while I’m talking (usually at almost exactly 5 minutes).

I’m using dahdi 2.5.0.1 with a TDM400P connected to PSTN.

The console doesn’t show anything ( dahdi 4 is PSTN):

How to unregister a sip trunk

Hi
In our asterisk we have more than 3 sip trunks.When we do “sip
reload” from CLI/AMI we need to unregister all 3 trunks and register
available trunks only. Now in asterisk unregisteration is not
happening.Please help on this

Thanks
NIkhil

ast_debug messages not showing up

Hi,

I’m running Asterisk 1.8.7.1 on Gentoo. I set `core set debug 9` but don’t
see any debug messages on the console. I do get the verbose messages from
ast_verbose. Is there something I need to configure to see these messages?

Regards,
Yahya