This is a hard one to explain. My home PSTN line is connected via an Openvox A400P card to my Asterisk 1.6.2.23 box which then routes incoming calls to my 2 SCCP extensions.

The calls are routed just fine, but when a call is answered at one of the extensions or externally (by a home telephone) the asterisk extensions continue to ring one more time. Is there a way to have Asterisk drop an incoming PSTN call as soon as it’s answered?

CLI output when receiving a PSTN call: Starting simple switch on ‘DAHDI/3-1′

Hi

What should be the optimal settings for HWEC related parameters in
wanpipe1.conf to be able to receive Fax over T.30 on a Sangoma A108DE Card ?

In particular, should I enable echo ? If yes, to what parameter ? What
other values like RXGAIN, TXGAIN should I tune ?

TDMV_HW_DTMF
= YES # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT = YES # YES: receive fax 1100hz
events from hardware
HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo
cancelation enabled with nlp (default)

# OCT_SPEECH: improves software tone detection by disabling NLP (echo
possible)

# OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions.
HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out
of incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on
the line – could break fax
HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables
acustic echo cancelation
HWEC_NLP_DISABLE = NO # NO: default YES: guarantees
software tone detection (possible echo)
HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx
audio level to be maintained (-20 default)
HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx
audio level to be maintained (-20 default)
HWEC_TX_GAIN = 0 # 0: disable
-24-24: db values to be applied to tx signal
HWEC_RX_GAIN = 0 # 0: disable
-24-24: db values to be applied to tx signal

Hello

We have a Genband C3 Switch and a couple of customers that operate asterisk
PBXes connected via SIP Trunk. All of them still use some 1.6.X asterisk and
this works fine.

One customer uses a 1.8 version and has a very strange problem:

Asterisk 1.8.10.0-1digium1~lucid built by pbuilder @ nighthawk on a x86_64
running Linux on 2012-03-06 01:51:21 UTC

When a call reaches this asterisk the call get’s dropped after a couple of
seconds with this message:

[Mar 27 13:47:35] WARNING[29806]: chan_sip.c:3641 retrans_pkt: Retransmission
timeout reached on transmission F47F6F67@7f33ff47 for seqno 907689273
(Critical Response) — See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Mar 27 13:47:35] WARNING[29806]: chan_sip.c:3670 retrans_pkt: Hanging up call
F47F6F67@7f33ff47 – no reply to our critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

Well, the Wiki is not very helpfull, as the problems explained there do not
seem to be the cause.

I sniffed the connection on the Asterisk Box itself and what I see is this
flow of SIP messages:

C3 < => Asterisk

=> Invite +sdp
< = 401 Unauthorized
=> Ack
=> Invite +sdp +auth
< = 100 Trying
< = 180 Ringing
< = RTP Audio
< = 200 OK +sdp
=> Ack
< => RTP Audio
< = 200 OK +sdp
=> Ack
< = 200 OK +sdp
=> Ack
< = 200 OK +sdp
=> Ack
< = Bye
=> Ack
< = Bye
=> Ack
< = Bye
=> Ack

Well apparently the C3, from my point of view sends correct Acks to the
Asterisk, but somehow the Asterisk ignores them.

I did try to get more information with debug and verbose level set to ’99′,
but I don’t see more messages

Does anyone have a clue, why acks could be not accepted by asterisk 1.8.10.0 ?

The other way round (asterisk => c3) the calls work fine.

Regards

Benoit Panizzon

I would like to fetch my extensions from the database. I created a dynamic
hint, but doesn’t seem to work. The BLF on my phone doesn’t change when the
state of the extension changed.

This is in my dialplan:

exten => _ZXX!,hint,${SIP_BYEXT(${EXTEN},${CONTEXT})}
exten => _ZXX!,1,Verbose(3, Search extension ${EXTEN} in context ${CONTEXT})
same => n,Set(SIP=${SIP_BYEXT(${EXTEN},${CONTEXT})})
same => n,GotoIf(${SIP}?:notFound)
same => n,SIPAddHeader(Alert-Info: internal)
same => n,Dial(${SIP})
same => n(notFound),Playback(you-dialed-wrong-number)
same => n,Hangup

Is something like this possible?

On 3/26/2012 1:11 PM, bilal ghayyad wrote:
> If it possible, then is it possible to be a configuration per user?

Just expanding on Jim’s answer-

to allow user “example” with password “secret” from 192.168.0.*, do
something like:

in /etc/asterisk/sip.conf:

[example]
type=friend
secret=secret
host=dynamic
deny=0.0.0.0/0
permit=192.168.0.0/24

then asterisk -rx “sip reload”