Page 385 - Asterisk™: The Future of Telephony
P. 385

advantages of DNS, whereas disabling them removes the ability to place SIP calls
                   based on domain names.

                              Currently, the support for SRV records in Asterisk is somewhat
                              lacking. If multiple SRV records are returned, Asterisk will use only
                              the first record.


                   Using  DNS  SRV  record  lookups  is  highly  recommended.  To  enable  them,  set
                   srvlookup=yes in the [general] section of sip.conf:
                       srvlookup=yes

               t1min
                   This  is  the  minimum  round-trip  time  for  messages  to  monitored  hosts  in
                   milliseconds. Defaults to 100 milliseconds:

                       t1min=100
               subscribecontext
                   Limits SUBSCRIBE requests to the specified context. This is useful if you want to
                   limit subscriptions to internal extensions, for example.
                   This option may also be set on a per-user or per-peer basis:
                       subscribecontext=internal
               t38pt_udptl
                   Setting t38pt_udptl to yes enables T.38 fax (UDPTL) passthrough on SIP-to-SIP
                   calls, provided both parties have T.38 support. This setting must be enabled in the
                   general section for all devices to work. You can then disable it on a per-device basis:

                       t38pt_udptl=yes|no
                              T.38 fax passthrough works only in SIP-to-SIP calls, without any
                              local or agent channel being used. Asterisk cannot currently orig-
                              inate or terminate T.38 fax calls; it can only passthrough UDPTL
                              from one device to another.


               tos_sip, tos_audio, and tos_video
                   Asterisk can set the TOS bits in the IP header to help improve performance on
                   routers  that  respect  TOS  bits  in  their  routing  calculations.  The  tos_sip,
                   tos_audio, and tos_video settings control the TOS bits for the SIP messages, the
                   RTP audio, and RTP video respectively. Valid: CS0, CS1, CS2, CS3, CS4, CS5, CS6, CS7,
                   AF11, AF12, AF13, AF21, AF22, AF23, AF31, AF32, AF33, AF41, AF42, AF43, and ef (expe-
                   dited forwarding). You may also use a numeric value for the TOS bits.
                   For more information, see the doc/ip-tos.txt file in the Asterisk source directory.






                                                                                 SIP | 357
   380   381   382   383   384   385   386   387   388   389   390