Hi all

im facing a problem with calls over a public IP , i can reach the other part but when he pick up i can't hear any voice ,and when i try to callthat number from outside it's showing as busy


Im new in the VOIP and i hope i can learn from you guys

this is the log calls


== Spawn extension (macro-exten-vm, s-CONGESTION, 2) exited non-zero on 'Zap/2-1' in macro 'ex ten-vm'
== Spawn extension (macro-exten-vm, s-CONGESTION, 2) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
-- Executing [39303901@from-internal:1] Macro("SIP/556-093bdef0", "user-callerid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/556-093bdef0", "user-callerid: device 556") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/556-093bdef0", "AMPUSER=556") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/556-093bdef0", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/556-093bdef0", "1|Set|REALCALLERIDNUM=556 ") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/556-093bdef0", "REALCALLERIDNUM is 556") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/556-093bdef0", "AMPUSER=556") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/556-093bdef0", "AMPUSERCIDNAME=Hussain") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/556-093bdef0", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/556-093bdef0", "AMPUSERCID=556") in new stac k
-- Executing [s@macro-user-callerid:10] Set("SIP/556-093bdef0", "CALLERID(all)="Hussain" <55 6>") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/556-093bdef0", "REALCALLERIDNUM=556") in ne w stack
-- Executing [s@macro-user-callerid:12] ExecIf("SIP/556-093bdef0", "0|Set|CHANNEL(language)= ") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/556-093bdef0", "TTL: ARG1: SKIPTTL") in n ew stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/556-093bdef0", "1?continue") in new stac k
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/556-093bdef0", "Using CallerID "Hussain" < 556>") in new stack
-- Executing [39303901@from-internal:2] Set("SIP/556-093bdef0", "_NODEST=") in new stack
-- Executing [39303901@from-internal:3] Macro("SIP/556-093bdef0", "record-enable|556|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/556-093bdef0", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/556-093bdef0", "recordingcheck|20090411-1128 34|1239438514.20") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca -- Executing [s@macro-record-enable:5] NoOp("SIP/556-093bdef0", "No recording needed") in ne elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca -- Executing [39303901@from-internal:4] Macro("SIP/556-093bdef0", "dialout-trunk|1|39303901| |") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/556-093bdef0", "DIAL_TRUNK=1") in new stack
-- Executing [s@macro-dialout-trunk:2] ExecIf("SIP/556-093bdef0", "0|Authenticate|") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/556-093bdef0", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/556-093bdef0", "DIAL_NUMBER=39303901") in ne w stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/556-093bdef0", "DIAL_TRUNK_OPTIONS=tr") in n ew stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/556-093bdef0", "GROUP()=OUT_1") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/556-093bdef0", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/556-093bdef0", "0?skipoutcid") in new sta ck
-- Executing [s@macro-dialout-trunk:10] Set("SIP/556-093bdef0", "DIAL_TRUNK_OPTIONS=") in ne w stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/556-093bdef0", "outbound-callerid|1") in new stack
-- Executing [s@macro-outbound-callerid:1] GotoIf("SIP/556-093bdef0", "1?start") in new stac k
-- Goto (macro-outbound-callerid,s,3)
-- Executing [s@macro-outbound-callerid:3] NoOp("SIP/556-093bdef0", "REALCALLERIDNUM is 556" ) in new stack
-- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/556-093bdef0", "1?normcid") in new st ack
-- Goto (macro-outbound-callerid,s,9)
-- Executing [s@macro-outbound-callerid:9] Set("SIP/556-093bdef0", "USEROUTCID=17313556") in new stack
-- Executing [s@macro-outbound-callerid:10] Set("SIP/556-093bdef0", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:11] Set("SIP/556-093bdef0", "TRUNKOUTCID=") in new s tack
-- Executing [s@macro-outbound-callerid:12] GotoIf("SIP/556-093bdef0", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing [s@macro-outbound-callerid:16] GotoIf("SIP/556-093bdef0", "1?usercid") in new s tack
-- Goto (macro-outbound-callerid,s,18)
-- Executing [s@macro-outbound-callerid:18] GotoIf("SIP/556-093bdef0", "0?report") in new st ack
-- Executing [s@macro-outbound-callerid:19] Set("SIP/556-093bdef0", "CALLERID(all)=17313556" ) in new stack
-- Executing [s@macro-outbound-callerid:20] GotoIf("SIP/556-093bdef0", "1?report:hidecid") i n new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing [s@macro-outbound-callerid:22] NoOp("SIP/556-093bdef0", "CallerID set to "" <17 313556>") in new stack
-- Executing [s@macro-dialout-trunk:12] AGI("SIP/556-093bdef0", "fixlocalprefix") in new sta ck
elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca -- Executing [s@macro-dialout-trunk:13] Set("SIP/556-093bdef0", "OUTNUM=39303901") in new st elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca -- Executing [s@macro-dialout-trunk:14] Set("SIP/556-093bdef0", "custom=ZAP/g0") in new stac elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca -- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/556-093bdef0", "1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/556-093bdef0", "dialout-trunk-predial-hoo k|") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/556-093bdef0", "0?bypass|1") in new stac k
-- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/556-093bdef0", "0?customtrunk") in new s tack
-- Executing [s@macro-dialout-trunk:20] Dial("SIP/556-093bdef0", "ZAP/g0/39303901|300|") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/39303901
elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/556-093bdef0' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/556-093bdef0'
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/556-093bdef0", "hangupcall|") in new stack
elastix*CLI> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/565-092328d0", "outbound-ca -- Executing [s@macro-hangupcall:2] NoCDR("SIP/556-093bdef0", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/556-093bdef0", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/556-093bdef0", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/556-093bdef0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/556-093bdef0", "") in new stack



cheers

if you're behind a firewall please read elastix without tears and check especially the following topics:

firewall setting and NAT port forwarding
externip and localip setting in your conf files
rtp setting in your conf files

the lack of voice could be explained by the lack of RTP port mapping
the unsuccessful inbound could be explained by the lack of SIP and RTP port mapping (assuming that the inbound routing is ok)

If all of that is ok and only inbound calls are still not working, in the general setting tab allow anonymous sip call and log in the CLI to check the trace and the DID presented by your provider to make it match with your inbound route.

Read up about the sip_nat.conf file, basically you need to tell it your public-IP or dynamic DNS domain name

Welcome to the Forums!

thanks patrick and chiling but im still facing the same problem i have added the IP on the sip_nat.conf and also added it on sip.conf but nothing have changed

i have add the following lines

nat=yes
externip=nnn.nnn.nnn.nnn
; externhost=<mydomain.com.au>­
localnet=192.168.1.0/255.255.0.0
externrefresh=10

but im not sure that if these lines correct !! can somebody tell me how can i add the public IP in the sip.conf file and what is the correct lines because i have copy these lines from the sip.conf trixbox !










Thanks all

cheers

but did you open the rtp ports on your router as explained in Elastix without tears (ch. 41.1)?

[quote]While most domestic routers using UPnP do not require port forwarding, some routers do. Even if your router does not require port forwarding for ELASTIX to work, do yourself a favour and forward a few of the relevant ports to your Asterisk IP address anyway

it's already open!

but i think there is a place where to define the public IP !

im not sure about that but do i have to define it in some files?

hello,

Another option could be the usage of an iax2 remote extension. IAX2 handles very well nat issues, if you are using it, you will need to enable/open port 4569 UDP. By the way there is no problem calling from iax2 to sip extensions and viceversa. The only thing that you will need to take into account is that the ip devices supports iax2, you may try using a softphone (for example zoiper)

the problem is that i can't use iax2 because the phone that im useing support sip only !

I saw that you are using analog lines, if you are using a sip trunk, on the outbound route, set it up to use the sip trunk instead of the analog lines.

I assume that the problem that you have is when you try to dialout using a remote extension you get that problem, what about if you try to do it with a local extension, what do you get. Also with your remote extension, have you tried to dial into voicemail (*98), just to make sure if you get audio, if you do, choose any option, just to make sure that asterisk is receiving audio