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

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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 3.5 years ago 2 Answers

Best Way to Send Fax on Sangoma ?

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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 3.5 years ago 0 Answers

Retransmission Timeout

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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:

Asterisk Users 3.5 years ago 0 Answers

Dynamic hint from db?

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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 3.5 years ago 1 Answer

username and password, also to be from the LAN

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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 3.5 years ago 0 Answers

in sip.conf or dialplan or db?

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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.
> .....
>…

Asterisk Users 3.5 years ago 1 Answer

GSM setup recommendations

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Hello list,
We are interested in configuring an Asterisk based GSM server in Africa
an I would like your recommendations. The requirements are:
1) A system with at least 8 GSM cards
2) 2 pstn lines from here (the US) will directly link with 2 lines on
the GSM server and the rest of the U.S. users will be routed to the
remaining GSM lines based on availability.
3) Web interface will also be needed for calls and text messages
*
Based on these requirements, what do you guys recommend?

Asterisk Users 3.5 years ago 0 Answers

Cdr Logs modify Disposition on Unsuccessful call

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Hi Team, I would like capture SS7 Error Code in CDRs, Specifically of outbound call
from the asterisk. calls generated using .call file. In extension.conf extens gets excuted on successful call only , So that on
h extension reason of hangup is captured. But i am not aware of any
provision that capture on Unsuccessful call. please guide on this or suggest any patch. Thanks
Vinod d

Asterisk Users 3.5 years ago 0 Answers