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

Thanks,

- D

hi folks,

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

thanks in advance

Hi folks.

I’ve got a problem dialing with my new Snom M9 via TLS on asterisk 1.8.7.1 .
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
https://issues.asterisk.org/jira/browse/ASTERISK-19003?focusedCommentId=186012#comment-186012

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 ).

Cheers,
Gregor.

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.

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,
Samuel

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.
>