Audio Problem

Hi all looking for help

I am using FreePBX 2.8.1.4 and I have couple of trunks working just fine i am trying this new trunk Inphonex and the calls seems to go though but there is no two way audio my trunk settings are as below, looked though all the forums and tried everything please help

canreinvite=no
context=from-varphonex
fromdomain=varphonex.com
fromuser=XXXXX
host=sip.varphonex.com
disallow=all
allow=ulaw,alaw,gsm
nat=yes
secret=XXXXX
externip=XXXXXXXX
insecure=invite
type=peer
username=XXXXXXX

and my CLI output is as below

[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:1] Macro(“SIP/105-00000452”, “user-callerid,SKIPTTL,”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:1] Set(“SIP/105-00000452”, “AMPUSER=105”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:2] GotoIf(“SIP/105-00000452”, “0?report”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:3] ExecIf(“SIP/105-00000452”, “1?Set(REALCALLERIDNUM=105)”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:4] Set(“SIP/105-00000452”, “AMPUSER=105”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:5] Set(“SIP/105-00000452”, “AMPUSERCIDNAME=Sydney Ipad”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:6] GotoIf(“SIP/105-00000452”, “0?report”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:7] Set(“SIP/105-00000452”, “AMPUSERCID=105”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:8] Set(“SIP/105-00000452”, “CALLERID(all)=“Sydney Ipad” <105>”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:9] ExecIf(“SIP/105-00000452”, “0?Set(CHANNEL(language)=)”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:10] GotoIf(“SIP/105-00000452”, “1?continue”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Goto (macro-user-callerid,s,19)
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:19] Set(“SIP/105-00000452”, “CALLERID(number)=105”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:20] Set(“SIP/105-00000452”, “CALLERID(name)=Sydney Ipad”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:21] NoOp(“SIP/105-00000452”, “Using CallerID “Sydney Ipad” <105>”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:2] NoOp(“SIP/105-00000452”, “Calling Out Route: inphonex”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:3] Set(“SIP/105-00000452”, “MOHCLASS=default”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:4] Set(“SIP/105-00000452”, “_NODEST=”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:5] Macro(“SIP/105-00000452”, “record-enable,105,OUT,”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/105-00000452”, “1?check”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Goto (macro-record-enable,s,4)
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:4] ExecIf(“SIP/105-00000452”, “0?MacroExit()”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:5] GotoIf(“SIP/105-00000452”, “0?Group:OUT”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Goto (macro-record-enable,s,15)
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:15] GotoIf(“SIP/105-00000452”, “0?IN”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:16] ExecIf(“SIP/105-00000452”, “1?MacroExit()”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:6] Macro(“SIP/105-00000452”, “dialout-trunk,7,61296715429,”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:1] Set(“SIP/105-00000452”, “DIAL_TRUNK=7”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:2] GosubIf(“SIP/105-00000452”, “0?sub-pincheck,s,1”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:3] GotoIf(“SIP/105-00000452”, “0?disabletrunk,1”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:4] Set(“SIP/105-00000452”, “DIAL_NUMBER=61296715429”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:5] Set(“SIP/105-00000452”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:6] Set(“SIP/105-00000452”, “OUTBOUND_GROUP=OUT_7”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:7] GotoIf(“SIP/105-00000452”, “1?nomax”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Goto (macro-dialout-trunk,s,9)
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:9] GotoIf(“SIP/105-00000452”, “0?skipoutcid”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:10] Set(“SIP/105-00000452”, “DIAL_TRUNK_OPTIONS=”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:11] Macro(“SIP/105-00000452”, “outbound-callerid,7”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:1] ExecIf(“SIP/105-00000452”, “0?Set(CALLERPRES()=)”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:2] ExecIf(“SIP/105-00000452”, “0?Set(REALCALLERIDNUM=105)”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:3] GotoIf(“SIP/105-00000452”, “1?normcid”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Goto (macro-outbound-callerid,s,6)
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:6] Set(“SIP/105-00000452”, “USEROUTCID=”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:7] Set(“SIP/105-00000452”, “EMERGENCYCID=”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:8] Set(“SIP/105-00000452”, “TRUNKOUTCID=”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:9] GotoIf(“SIP/105-00000452”, “1?trunkcid”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Goto (macro-outbound-callerid,s,12)
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:12] ExecIf(“SIP/105-00000452”, “0?Set(CALLERID(all)=)”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:13] ExecIf(“SIP/105-00000452”, “0?Set(CALLERID(all)=)”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:14] ExecIf(“SIP/105-00000452”, “0?Set(CALLERID(all)=)”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:15] ExecIf(“SIP/105-00000452”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:12] GosubIf(“SIP/105-00000452”, “1?sub-flp-7,s,1”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:1] ExecIf(“SIP/105-00000452”, “1?Return()”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:13] Set(“SIP/105-00000452”, “OUTNUM=01161296715429”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:14] Set(“SIP/105-00000452”, “custom=SIP/out_varphonex”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:15] ExecIf(“SIP/105-00000452”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:16] Macro(“SIP/105-00000452”, “dialout-trunk-predial-hook,”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]predial-hook:1] MacroExit(“SIP/105-00000452”, “”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:17] GotoIf(“SIP/105-00000452”, “0?bypass,1”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:18] GotoIf(“SIP/105-00000452”, “0?customtrunk”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] pbx.c: – Executing [[email protected]:19] Dial(“SIP/105-00000452”, “SIP/out_varphonex/01161296715429,300,”) in new stack
[Jan 22 16:31:16] VERBOSE[25258] netsock2.c: == Using SIP RTP TOS bits 184
[Jan 22 16:31:16] VERBOSE[25258] netsock2.c: == Using SIP RTP CoS mark 5
[Jan 22 16:31:16] VERBOSE[25258] app_dial.c: – Called SIP/out_varphonex/01161296715429
[Jan 22 16:31:19] VERBOSE[25258] app_dial.c: – SIP/out_varphonex-00000453 is ringing
[Jan 22 16:31:19] VERBOSE[25258] app_dial.c: – SIP/out_varphonex-00000453 is making progress passing it to SIP/105-00000452
[Jan 22 16:31:23] VERBOSE[25258] app_dial.c: – SIP/out_varphonex-00000453 answered SIP/105-00000452
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:1] Macro(“SIP/105-00000452”, “hangupcall,”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/105-00000452”, “1?endmixmoncheck”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Goto (macro-hangupcall,s,9)
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:9] NoOp(“SIP/105-00000452”, “End of MIXMON check”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:10] GotoIf(“SIP/105-00000452”, “1?nomeetmemon”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Goto (macro-hangupcall,s,28)
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:28] NoOp(“SIP/105-00000452”, “End of MEETME check”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:29] GotoIf(“SIP/105-00000452”, “1?noautomon”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Goto (macro-hangupcall,s,34)
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:34] NoOp(“SIP/105-00000452”, “TOUCH_MONITOR_OUTPUT=”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:35] GotoIf(“SIP/105-00000452”, “1?noautomon2”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Goto (macro-hangupcall,s,41)
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:41] NoOp(“SIP/105-00000452”, “MONITOR_FILENAME=”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:42] GotoIf(“SIP/105-00000452”, “1?skiprg”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Goto (macro-hangupcall,s,45)
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:45] GotoIf(“SIP/105-00000452”, “1?skipblkvm”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Goto (macro-hangupcall,s,48)
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:48] GotoIf(“SIP/105-00000452”, “1?theend”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Goto (macro-hangupcall,s,50)
[Jan 22 16:31:28] VERBOSE[25258] pbx.c: – Executing [[email protected]:50] AGI(“SIP/105-00000452”, “hangup.agi”) in new stack
[Jan 22 16:31:28] VERBOSE[25258] res_agi.c: – Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
[Jan 22 16:31:29] VERBOSE[25258] res_agi.c: – <SIP/105-00000452>AGI Script hangup.agi completed, returning 0
[Jan 22 16:31:29] VERBOSE[25258] pbx.c: – Executing [[email protected]:51] Hangup(“SIP/105-00000452”, “”) in new stack
[Jan 22 16:31:29] VERBOSE[25258] app_macro.c: == Spawn extension (macro-hangupcall, s, 51) exited non-zero on ‘SIP/105-00000452’ in macro ‘hangupcall’
[Jan 22 16:31:29] VERBOSE[25258] features.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/105-00000452’
[Jan 22 16:31:29] VERBOSE[25258] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/105-00000452’ in macro ‘dialout-trunk’
[Jan 22 16:31:29] VERBOSE[25258] pbx.c: == Spawn extension (from-internal, 361296715429, 6) exited non-zero on ‘SIP/105-00000452’

Nothing relevent in the log. Do you have your SIP NAT settings configured in the SIP Settings module and did you forward UDP 5060 and 10,000-20,000 in your router/firewall?

You can reduce the number of RTP ports to 4*max concurrent calls in /etc/asterisk/rtp.conf

When opening SIP to the world make sure you follow security best practices, intruse detection and strong secrets.

Hi thanks for your reply… yes i have the sip NAT configured I can make calls via other trunk but no audio in this one though the call is being connected made test calls when there were no traffic but still the same problem

i would think this is a firewall issue.

I think the ISP has blocked that certain IP other trunks are working