* You are viewing Posts Tagged ‘hi folks’

app_swift beta release

Hi folks,

Just a note to let everyone know I’ve finally finished up the new BETA release of app_swift (now v3.0.1 b1).

This release introduces some pretty major changes to app_swift such as:

– The entire code-base has now been unified and the build system auto detects which Asterisk version you’re using (yay! one branch!)

– Auto-detection and support for both the Cepstral 5.0 and 6.0 engines

– Support for Asterisk 1.4, 1.6, 1.8, 1.8 Certified, and 10

– Asterisk 1.2 support has been dropped.

I have only been able to do some basic testing with all these permutations of Asterisk and the Cepstral engines on a few of my machines here at the house and need some volunteers to help out and be guinea-pigs.

Please email me directly with any feedback you might have.

I’ve updated my github repo with the new app_swift code which can be downloaded using git.

git clone git://github.com/dmsessions/app_swift.git


– D

a2billing script

hi folks,

i was wondering if some one has a2billing script,which can be used to
install a2billing easly ?

thanks in advance

TLS bug in asterisk?

Hi folks.

I’ve got a problem dialing with my new Snom M9 via TLS on asterisk .
Registration works like a charm – the phone becomes ‘AVAILABLE’.
An INVITE is responded by a 401 to be expected, but then asterisk closes the TLS connection upon the Snom’s ACK.

The authenticated INVITE the Snom sends immediately after the ACK meets a closed socket and merely triggers a TCP RST packet on asterisk’s behalf.

There’s no ERROR or WARNING put out on the asterisk CLI.
The only hint I get is asterisk complaining about not finding the CSeq anymore it used a second ago for the beginning of the dialog.

I couldn’t really figure a reason for asterisk to close the connection when it should wait for an authenticated INVITE, so I posted the problem details in the bug tracker under

I’d be very happy though, if someone could show me that this is not a bug, or how to work around it (I’ve got the Snom for about one more week, and then I’ll have to decide whether to return it ;) ).


skype connect & early media

hi folks.

when i use regular PSTN(sip phone -> asterisk -> PRI) to call
certain numbers and when that number is unavailable. i usually
hear an early media message saying “blahblah is unavailable,
please try again”. but when i use skype connect(sip phone -> asterisk
-> skype connect). i just hear ring back tone for about 20 seconds
and then become fast busy. is there any setting i’m unaware of
when setting up sip w/ skype connect?

any suggestions would be appreciated.

forwarding early media

Hi folks,

Please discard the e-mail since it seems to be a problem of the E1 provider
and nothign related to asterisk.

Apologies for the noise,

On 17 October 2011 18:32, samuel wrote:

> Hi folks,
> I’m having an issue with an asterisk 1.4.36 with an E1 card that is not
> forwarfing the early media a remote SIP end-point is creating.
> –incoming E1 call–>asterisk 1.4.36—->SIP endpoint (which happens to be
> an asterisk 1.6.20).
> I’ve checked signalling and the remote end-point returns 183 with the
> correct SDP. In the asterisk 1.4.36 I have progressinband=yes to precissely
> enable this feature and debugging from the asterisk console (both sip and
> rtp), the asterisk gateway gets the 183, create the remote peer and starts
> receiving the RTP (RTP From…blabnlabal).
> The only missing part is that the asterisk 1.4.36 instance gets the RTP
> audio from the SIP endpoint and forwards it to the E1 card.
> I’ve also played with prematuremedia parameter but got no change in the
> behaviour.
> Can anyone provide any hint about this issue? Both links to documentation
> and help debugging this issue will be highly appreciated.
> Thank you very much in advance,
> Samuel.

Digium FFA + Gafachi T38 outgoing issues

Hi, folks.

I’m having a heck of a time trying to get outgoing T38 faxing (I don’t
need inbound right now) working with FFA and Gafachi. G711 faxing works
(as well as can be expected over the internet), but I want the higher
reliability of T38.

I’m running Asterisk 10-beta1.

When I drop my callfile in to make the call, I get this: