How to stop ringing when incoming PSTN call is answered externally?

Report
Question

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'

Asterisk Users 3.3 years ago 2 Answer

Best Way to Send Fax on Sangoma ?

Report
Question

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

Asterisk Users 3.3 years ago 0 Answer

Retransmission timeout

Report
Question

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…

Asterisk Users 3.3 years ago 0 Answer

Dynamic hint from db?

Report
Question

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?

Asterisk Users 3.3 years ago 1 Answer

username and password, also to be from the LAN

Report
Question

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"

Asterisk Users 3.3 years ago 0 Answer