From 23a27a03100295fb43477ee4dd5f62ab1c9746d5 Mon Sep 17 00:00:00 2001 From: Thomas Ries Date: Sun, 13 May 2007 16:32:24 +0000 Subject: [PATCH] *** empty log message *** --- doc/asterisk_sample.txt | 236 ---------------------------------------- 1 file changed, 236 deletions(-) delete mode 100644 doc/asterisk_sample.txt diff --git a/doc/asterisk_sample.txt b/doc/asterisk_sample.txt deleted file mode 100644 index 6cd3760..0000000 --- a/doc/asterisk_sample.txt +++ /dev/null @@ -1,236 +0,0 @@ -Example Setup, Asterisk running on the same machine as siproxd (= NAT host). - -I use a iptables rule to redirect all outgoing SIP traffix from Asterisk -to siproxd. - -/etc/sysconfig/iptables: ------------------------- -*nat -###################################################################### -# NAT: redirect locally generated packets -:OUTPUT - [0:0] -######################### -# -# Asterisk Traffix via local siproxd. (must use DNAT to inbound IF! not REDIRECT) --A OUTPUT -o ppp+ -p udp --sport 5061 -j DNAT --to-destination 192.168.1.1:5060 - -COMMIT - - - - -/etc/asterisk/sip.conf: ------------------------ -[general] -context = default -allowoverlap = no ; Disable overlap dialing support. (Default is yes) -bindport = 5061 ; use a different port than 5060, as that port will be - ; occupied by siproxd! -bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) -srvlookup = yes ; Enable DNS SRV lookups on outbound calls - -; g726 sounds very bad, useless! -; g722 is a dead end, no conversion from/to possible -; g729 sound like through a long metal tube -disallow = all -allow = gsm,ulaw,alaw,adpcm,speex,g729,g723 -autoframing = yes - -allowexternaldomains = yes -allowexternalinvites = yes -allowguest = yes -allowsubscribe = no -allowtransfer = yes -alwaysauthreject = no -autodomain = yes -callevents = no -compactheaders = no -dumphistory = no -g726nonstandard = no -ignoreregexpire = no -jbenable = no -jbforce = no -jblog = no -maxcallbitrate = 384 -maxexpiry = 3600 -minexpiry = 180 -notifyringing = no -pedantic = no -promiscredir = no -recordhistory = no -relaxdtmf = no -rtcachefriends = no -rtsavesysname = no -rtupdate = no -sendrpid = yes -sipdebug = no -t1min = 100 -progressinband = no -t38pt_udptl = no -trustrpid = no -usereqphone = no -videosupport = no -; -; the following is required when using siproxd with local DNAT rule -nat=never -externip=192.168.1.1 -; -localnet = 192.168.0.0/16 ; my inbound network with local UAs -domain = 192.168.1.1 ; inbound IP of host running Asterisk and siproxd -domain = mynatfirewall ; -"- -canreinvite = no - -useragent = PBX ; sipcall.ch (and others?) require UA string - ; to be different from "AsteriskPBX" - -[authentication] -;---end--- - - -/etc/asterisk/users.conf ------------------------- -[general] -; -; Full name of a user -; -fullname = New User -userbase = 200 -; -; Create voicemail mailbox and use use macro-stdexten -; -hasvoicemail = yes -; -; Set voicemail mailbox 6000 password to 1234 -; -vmsecret = 1234 -; -; Create SIP Peer -; -hassip = yes -hasiax = no -; -; -; Create manager entry -; -hasmanager = no -; -; Remaining options are not specific to users.conf entries but are general. -; -callwaiting = yes -threewaycalling = yes -callwaitingcallerid = yes -transfer = yes -canpark = yes -cancallforward = yes -callreturn = yes -callgroup = 1 -pickupgroup = 1 -host = dynamic -localextenlength = 3 -allow_aliasextns = no -allow_an_extns = no -hasagent = no -hasdirectory = no - -;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; -; Local SIP UAs -;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; -[201] -callwaiting = yes -cid_number = 201 -context = local_sip -email = email@host.xx -fullname = Full Name -group = -hasagent = yes -hasdirectory = yes -hasiax = no -hasmanager = no -hassip = yes -hasvoicemail = yes -host = dynamic -mailbox = 201 -secret = -threewaycalling = yes -zapchan = -registeriax = no -registersip = yes -vmsecret = - -[202] -callwaiting = yes -cid_number = 202 -context = local_sip -email = email@host.xx -fullname = Full Name -group = -hasagent = yes -hasdirectory = yes -hasiax = no -hasmanager = no -hassip = yes -hasvoicemail = yes -host = dynamic -mailbox = 202 -secret = -threewaycalling = yes -zapchan = -registeriax = no -registersip = yes -vmsecret = - - -;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; -; SIP Trunks -;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; -; sipphone.com -[trunk_1] -disallow = all -allow = gsm,ulaw,alaw,adpcm,speex,g729,g723 -callerid = -contact = 1747669xxxx -context = DID_trunk_1 -dialformat = ${EXTEN:1} -fromdomain = proxy01.sipphone.com -fromuser = 1747669xxxx -group = -hasexten = no -hasiax = no -hassip = yes -host = proxy01.sipphone.com -insecure = very -port = 5060 -provider = -registeriax = no -registersip = yes -secret = -trunkname = Custom - sipphone1341 -trunkstyle = customvoip -username = 1747669xxxx - -; sipcall.ch -[trunk_3] -disallow = all -allow = gsm,ulaw,alaw,adpcm,speex,g729,g723 -callerid = -contact = 4132511xxxx -context = DID_trunk_3 -dialformat = ${EXTEN:1} -fromdomain = sip.backbone.ch -fromuser = 4132511xxxx -group = -hasexten = no -hasiax = no -hassip = yes -host = sip.backbone.ch -insecure = very -port = 5060 -provider = -registeriax = no -registersip = yes -secret = -trunkname = Custom - sipcall -trunkstyle = customvoip -username = 4132511xxxx - -;---end---