search for: sip_general_additional

...etc/asterisk/features_featuremap_custom.conf': == Found -- Added extension '70' priority 1 to parkedcalls (0xa8798b0) -- Reloading module 'res_phoneprov' (HTTP Phone Provisioning) == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/sip_general_additional.conf': == Found == Parsing '/etc/asterisk/sip_general_custom.conf': == Found == Parsing '/etc/asterisk/sip_nat.conf': == Found == Parsing '/etc/asterisk/sip_registrations_custom.conf': == Found == Parsing '/etc/asterisk/sip_registrations.conf':...

...mber (the second trunk) of the asterisk installation was used! The only difference between those two trunks is: The first trunk is configured to a ring group - the second trunk is configured directly to an extension. My solution after long time of digging around: I added progressinband=never to sip_general_additional.conf But this solution confuses me, because http://www.voip-info.org/wiki/view/Asterisk+sip+progressinband tells: progressinband=never Whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behaviour of Asterisk. ^^^...

...s. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten this set-up (Asterisk11 with Snom870s using TLS) to work and if so could you provide the details? I have this in Asterisk sip.conf (loaded through FreePBXs sip_general_additional.conf). tcpenable=yes tlsenable=yes tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt tlscafile=/etc/pki/tls/certs/ca-bundle.crt tlsdontverifyserver=yes tlscipher=ALL tlsclientmethod=tlsv1 And I have this for the test device context: [41712] deny=0.0.0.0/0.0.0.0 secret=NearlyANast...

...nnect my Softphone via VPN to Asterisk I'm registered and It's show via "pjsip list contacts" Then I try to call an internal number / other extension I get the following: "SIP/2.0 401 Unauthorized". The VPN net is list in pjsip.transports.conf:local_net=10.8.0.0/24 and sip_general_additional.conf:localnet=10.8.0.0/24 I'm not sure whats wrong here. Best Regards

...BB> tlsdontverifyserver=yes > JBB> tlscipher=ALL > JBB> tlsclientmethod=tlsv1 > > You are missing the tls key. > > The config name is tlsprivatekey; set that to the filename of your tls > key, akin to how tlscertfile is set. > > -JimC Thank you. The settings in sip_general_additional.conf are now: tcpenable=yes tlsenable=yes tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.pem tlscafile=/etc/pki/tls/certs/ca-bundle.crt tlsdontverifyserver=yes tlscipher=ALL tlsclientmethod=tlsv1 tlsprivatekey=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.key However, issuing...

.... > > Since this is an extraordinarily (to me) Byzantine environemnt I am > going to ask if any of you have gotten this set-up (Asterisk11 with > Snom870s using TLS) to work and if so could you provide the details? > > I have this in Asterisk sip.conf (loaded through FreePBXs > sip_general_additional.conf). > > tcpenable=yes > tlsenable=yes > tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt > tlscafile=/etc/pki/tls/certs/ca-bundle.crt > tlsdontverifyserver=yes > tlscipher=ALL > tlsclientmethod=tlsv1 > > And I have this for the test device context:...

...rver (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf ;--------------------------------------------------------------------------------; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make ; ; custom modifications, details at: http://fr...