Re: [asterisk-users] PJSIP logging fails
Did you try setting the debug verbosity to a number > 3? Alternatively, if you want to see a register packet, try running wireshark on the server and capture the request packets. Sree On 04/12/2017 08:55 PM, Saint Michael wrote: > I am trying to log my SIP registration attempts. > PJSIP is in logger mode, and I can see INVITES comingh, my SIP Register > does not show, especially the packet I send. > The only thing shown is: > res_pjsip_outbound_registration.c: No response received from 'snet' on > registration attempt to 'sip:7866314772-xnet', retrying in '60' > How do I see my own packet? > > > smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Logging endpoint IP address when PJSIP registration fails
Hi! I am seeing a lot of warnings of these types: res_pjsip_registrar.c: AOR '31' not found for endpoint 'anonymous' I am guessing these are coming from a scanner trying to scan for the extensions on the asterisk server. Is there any way to print the IP address of the endpoint trying to register an extension using PJSIP in asterisk 13? I can then configure fail2ban to temporarily hinder the scan. Regards, Sree -- _ -- 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] Optimizing forwarded and redirected calls by avoiding signaling and media data redirection
Hi! Apparently this is possible; my asterisk server is doing this when my SIP phone redirects the call with a SIP REFER message. The phone is excluded from the call after it transfers the call. I'll contact my ITSP if their trunk can also do this. Regards, Sree On 03/09/2017 11:03 PM, Sree Harsha Totakura wrote: > Hi! > > I'm having a setup where my asterisk PBX connects to PSTN via a single > SIP trunk. Now, when I transfer or redirect incoming calls from the SIP > trunk to another number which is routed through the SIP trunk, my > asterisk stays on the way; it just dials out the new destination number > the call is transferred/redirected to and connects the newly dialed > channel to the existing incoming channel. > > Since these two channels are in the same SIP trunk, would it be possible > to tell the trunk SIP server to not involve my asterisk anymore, both > for signaling and media data? Or is this inherently not possible via SIP? > > Regards, > Sree > -- _ -- 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] Optimizing forwarded and redirected calls by avoiding signaling and media data redirection
Hi! I'm having a setup where my asterisk PBX connects to PSTN via a single SIP trunk. Now, when I transfer or redirect incoming calls from the SIP trunk to another number which is routed through the SIP trunk, my asterisk stays on the way; it just dials out the new destination number the call is transferred/redirected to and connects the newly dialed channel to the existing incoming channel. Since these two channels are in the same SIP trunk, would it be possible to tell the trunk SIP server to not involve my asterisk anymore, both for signaling and media data? Or is this inherently not possible via SIP? Regards, Sree -- _ -- 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
