Re: [asterisk-users] What does pct mean?
Hi, Lost percent (%) Thanks, Carsten Am 11.02.2018 19:27 schrieb "Michael Maier" : > Hello, > > could somebody please tell me the meaning of "Pct" as seen in asterisk cli: > > ...Receive. .Transmit.. > CountLost Pct Jitter CountLost Pct Jitter RTT > > > Thanks, > Michael > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What does pct mean?
Hello, could somebody please tell me the meaning of "Pct" as seen in asterisk cli: ...Receive. .Transmit.. CountLost Pct Jitter CountLost Pct Jitter RTT Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add SNMP support to packaged asterisk on Debian stretch
On Fri, Feb 09, 2018 at 03:55:40PM +0100, Olivier wrote: > Hello, > > If I'm not mistaken SNMP support is missing in Debian Stretch packaged > Asterisk while this support is present in either Jessie or Buster (looking > at [1] or equivalent pages). > > Is it something that can be worked around or shall I fear a major obstacle > when re-packaging my own asterisk package for Stretch ? >From the changelog (linked fro https://packages.debian.org/stretch/asterisk-modules ) * Disable SNMP support for now libsnmp-dev pulls in libssl1.0-dev, which is not coinstallable with libssl-dev needed by Asterisk and all other dependencies I don't remember how difficult it would be to work around. In the worst case, try backporting libsnmp from Buster. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Progress audio associated with 180 Ringing not passed to extension when using pjsip
I’m setting up a new PBX in the Google cloud running FreePBX 14.0.1.36 / Asterisk 14.7.5. Most calls are fine, but when calling an AT&T landline that is busy, ringback tone is heard instead of the expected busy signal. An example of a failing number is +1 408 269 1999 (a test number that is always busy). My production system running FreePBX 2.11.0.43 / Asterisk 11.4.0 using chan_sip and the same phone and trunk does not have this problem. Both extension and AnveoDirect trunk are using pjsip. I’m using Anveo’s ‘Smart Route Option’ which sends the call directly to AT&T (no intermediate tandem carrier). I don’t understand why, but AT&T’s response to the INVITE is 180 Ringing with associated SDP. RTP (containing audio for busy signal) starts coming in, but the 180 Ringing passed to the extension does not have associated SDP so the phone continues to play ringback. If I change the trunk to use chan_sip instead, the problem disappears. There is also no problem if I force Anveo to send the call to Verizon (which acts as a tandem carrier for this call) ; Verizon sends 183 Progress with the busy signal audio. Tests with Flowroute did show the trouble (I assume that they would send this call directly to AT&T). I’d like to migrate to pjsip – is there a trunk setting or manual config edit that works around this issue? Or is it somehow related to my build and does not affect other users? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
