I want a call flow as follows:
UE1 -> ASterisk -> AppServ(B2BUA) -> ASterisk -> UE2
my flow is correct upto 200 OK but as soon as ACK is sent , ASterisk sends another INVITE(native bridging) towards UE and AS.
I donot want this extra INVITE to be sent - how can I disable native bridging done by asterisk.
sip configuration and dialplan is as follows:
[quote]
sip.conf:
[general]
sendrpid = yes
[919811098110]
type=friend
context=phones
host=dynamic
[919891300300]
type=friend
host=dynamic
context=phones [/quote]
[quote]==================
extension.conf
[globals]
[general]
[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()
[incoming_calls]
[internal]
exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()
exten => 919891300300,1,Verbose(1|Extension 919891300300)
exten => 919891300300,n,SET(CONTACT=$[${SIP_HEADER(Contact):0:4}])
exten => 919891300300,n,Verbose(1|Contact is ${CONTACT})
exten => 919891300300,n,GotoIf($[${CONTACT} = user]?vcc:vcc_1)
exten => 919891300300,n(vcc),SIPAddHeader(Route: sip:444@172.16.105.35:5060\;lr)
exten => 919891300300,n,SIPAddHeader(Route: sip:444@172.31.118.53:5060\;lr)
exten => 919891300300,n,Dial(SIP/919891300300@172.31.118.53:5060)
exten => 919891300300,n,Hangup()
exten => 919891300300,n(vcc_1),Dial(SIP/919891300300@10.203.154.137:7062)
exten => 919891300300,n,Hangup()
exten => 1000,1,Verbose(1|Extension 1000)
exten => 1000,n,Dial(SIP/1000,30)
exten => 1000,n,Hangup()
[user]
type=friend
host=172.31.118.53
fromuser=user
secret=my_special_secret
context=incoming_calls
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=ulaw
insecure=invite
[phones]
include => internal [/quote]
[quote]===============
console logs
– Executing [919891300300@phones:1] Verbose(“SIP/919811098110-006f4f00”, “1|Extension 919891300300”) in new stack
Extension 919891300300
– Executing [919891300300@phones:2] Set(“SIP/919811098110-006f4f00”, “CONTACT=user”) in new stack
– Executing [919891300300@phones:3] Verbose(“SIP/919811098110-006f4f00”, “1|Contact is user”) in new stack
Contact is user
– Executing [919891300300@phones:4] GotoIf(“SIP/919811098110-006f4f00”, “1?vcc:vcc_1”) in new stack
– Goto (phones,919891300300,5)
– Executing [919891300300@phones:5] SIPAddHeader(“SIP/919811098110-006f4f00”, “Route: sip:444@172.16.105.35:5060;lr”) in new stack
– Executing [919891300300@phones:6] SIPAddHeader(“SIP/919811098110-006f4f00”, “Route: sip:444@172.31.118.53:5060;lr”) in new stack
– Executing [919891300300@phones:7] Dial(“SIP/919811098110-006f4f00”, “SIP/919891300300@172.31.118.53:5060”) in new stack
– Called 919891300300@172.31.118.53:5060
– Executing [919891300300@phones:1] Verbose(“SIP/919811098110-007062b0”, “1|Extension 919891300300”) in new stack
Extension 919891300300
[Jan 7 19:07:27] WARNING[31387]: ast_expr2.fl:398 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting ‘-’ or ‘!’ or ‘(’ or ‘’; Input:
sip:
^
[Jan 7 19:07:27] WARNING[31387]: ast_expr2.fl:402 ast_yyerror: If you have questions, please refer to doc/channelvariables.txt in the asterisk source.
– Executing [919891300300@phones:2] Set(“SIP/919811098110-007062b0”, “CONTACT=0”) in new stack
– Executing [919891300300@phones:3] Verbose(“SIP/919811098110-007062b0”, “1|Contact is 0”) in new stack
Contact is 0CLI>
– Executing [919891300300@phones:4] GotoIf(“SIP/919811098110-007062b0”, “0?vcc:vcc_1”) in new stack
– Goto (phones,919891300300,9)
– Executing [919891300300@phones:9] Dial(“SIP/919811098110-007062b0”, “SIP/919891300300@10.203.154.137:7062”) in new stack
– Called 919891300300@10.203.154.137:7062
– SIP/10.203.154.137:7062-0070c7b0 is ringing
– SIP/172.31.118.53:5060-006fdd10 is ringing
– SIP/10.203.154.137:7062-0070c7b0 answered SIP/919811098110-007062b0
– Native bridging SIP/919811098110-007062b0 and SIP/10.203.154.137:7062-0070c7b0
– SIP/172.31.118.53:5060-006fdd10 answered SIP/919811098110-006f4f00
– Native bridging SIP/919811098110-006f4f00 and SIP/172.31.118.53:5060-006fdd10[/quote]