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…
I've got a problem dialing with my new Snom M9 via TLS on asterisk 184.108.40.206 .
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…
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.
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
> 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…