problema dial plan generado por FREEPBX en asterisk

buenos dias

he estado intentado instalar un servidor asterisk en debian leny

instale debian leny 5.0 + asterisk 1.4 + freepbx 2.5 todo se instalo muy bien

aparentemente.

cree dos extensiones sip ..una extension 200 y otra 300 intento hacer un llamada entre estso dos softphone xlite y me dice error 404 no se puede hacer la llamada..SI llamo al voicemail o a cualquier cosa del dialplan lo hace pero llamadas no las realiza :S. lo primero que una piensa es bueno debe haber algun problema en sip.conf pero decidi observar el debug y ver que arroja sip.

y me dice lo siguiente:

v=0
o=- 6 2 IN IP4 192.168.1.14
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.14
t=0 0
m=audio 28326 RTP/AVP 107 0 8 18 101
a=alt:1 3 : duP+tm9U 1QHTrAtU 192.168.1.14 28326
a=alt:2 2 : Q7KuF8Nz UD3CFYNs 192.168.74.1 28326
a=alt:3 1 : 3sTVPIDX MmeyBfKJ 192.168.180.1 28326
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (13 headers 15 lines) ---
Sending to 192.168.1.14 : 50426 (NAT)
Using INVITE request as basis request - MWUzNGZkNDE4OGU4YjdmYTg1ZTQ4YjA1ZTI3YTEwMGY.
Found user '200'
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found unknown media description format BV32 for ID 107
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.14:28326
Looking for 300 in from-internal (domain 192.168.1.5)
list_route: hop:

<--- Transmitting (NAT) to 192.168.1.14:50426 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.14:50426;branch=z9hG4bK-d8754z-b96c56174d467f7e-1---d8754z-;received=192.168.1.14;rport=50426
From: "200";tag=1b02126e
To: "300"
Call-ID: MWUzNGZkNDE4OGU4YjdmYTg1ZTQ4YjA1ZTI3YTEwMGY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact:
Content-Length: 0

<------------>
-- Executing [300@from-internal:1] Macro("SIP/200-00000007", "exten-vm|300|300") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/200-00000007", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/200-00000007", "user-callerid: device 200") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/200-00000007", "AMPUSER=200") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/200-00000007", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/200-00000007", "1|Set|REALCALLERIDNUM=200") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/200-00000007", "REALCALLERIDNUM is 200") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/200-00000007", "AMPUSER=200") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/200-00000007", "AMPUSERCIDNAME=200") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/200-00000007", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/200-00000007", "AMPUSERCID=200") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/200-00000007", "CALLERID(all)="200" <200>") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/200-00000007", "REALCALLERIDNUM=200") in new stack
-- Executing [s@macro-user-callerid:12] ExecIf("SIP/200-00000007", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/200-00000007", "TTL: ARG1: 300") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/200-00000007", "0?continue") in new stack
-- Executing [s@macro-user-callerid:15] Set("SIP/200-00000007", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("SIP/200-00000007", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/200-00000007", "Using CallerID "200" <200>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/200-00000007", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/200-00000007", "VMBOX=300") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/200-00000007", "EXTTOCALL=300") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/200-00000007", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/200-00000007", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/200-00000007", "RT=15") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/200-00000007", "record-enable|300|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/200-00000007", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/200-00000007", "recordingcheck|20100726-105730|1280158050.7") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/200-00000007", "No recording needed") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/200-00000007", "dial|15|tr|300") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/200-00000007", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/200-00000007", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:4] NoOp("SIP/200-00000007", "Returned from dialparties with no extensions to call and DIALSTATUS: ") in new stack
-- Executing [s@macro-exten-vm:10] Set("SIP/200-00000007", "SV_DIALSTATUS=") in new stack
-- Executing [s@macro-exten-vm:11] GosubIf("SIP/200-00000007", "0?docfu|1") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/200-00000007", "0?docfb|1") in new stack
-- Executing [s@macro-exten-vm:13] Set("SIP/200-00000007", "DIALSTATUS=") in new stack
-- Executing [s@macro-exten-vm:14] NoOp("SIP/200-00000007", "Voicemail is 300") in new stack
-- Executing [s@macro-exten-vm:15] GotoIf("SIP/200-00000007", "0?s-|1") in new stack
-- Executing [s@macro-exten-vm:16] NoOp("SIP/200-00000007", "Sending to Voicemail box 300") in new stack
-- Executing [s@macro-exten-vm:17] Macro("SIP/200-00000007", "vm|300|") in new stack
-- Executing [s@macro-vm:1] Macro("SIP/200-00000007", "user-callerid|SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/200-00000007", "user-callerid: 200 200") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/200-00000007", "AMPUSER=200") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/200-00000007", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/200-00000007", "0|Set|REALCALLERIDNUM=200") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/200-00000007", "REALCALLERIDNUM is 200") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/200-00000007", "AMPUSER=200") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/200-00000007", "AMPUSERCIDNAME=200") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/200-00000007", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/200-00000007", "AMPUSERCID=200") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/200-00000007", "CALLERID(all)="200" <200>") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/200-00000007", "REALCALLERIDNUM=200") in new stack
-- Executing [s@macro-user-callerid:12] ExecIf("SIP/200-00000007", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/200-00000007", "TTL: 64 ARG1: SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/200-00000007", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/200-00000007", "Using CallerID "200" <200>") in new stack
-- Executing [s@macro-vm:2] Set("SIP/200-00000007", "VMGAIN=""") in new stack
-- Executing [s@macro-vm:3] GotoIf("SIP/200-00000007", "1?vmx|1") in new stack
-- Goto (macro-vm,vmx,1)
-- Executing [vmx@macro-vm:1] GotoIf("SIP/200-00000007", "0?s-|1") in new stack
-- Executing [vmx@macro-vm:2] Set("SIP/200-00000007", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:3] GotoIf("SIP/200-00000007", "1?notdirect") in new stack
-- Goto (macro-vm,vmx,5)
-- Executing [vmx@macro-vm:5] NoOp("SIP/200-00000007", "Checking if ext 300 is enabled: ") in new stack
-- Executing [vmx@macro-vm:6] GotoIf("SIP/200-00000007", "1?s-|1") in new stack
-- Goto (macro-vm,s-,1)
-- Executing [300@from-internal:2] Hangup("SIP/200-00000007", "") in new stack
== Spawn extension (from-internal, 300, 2) exited non-zero on 'SIP/200-00000007'
-- Executing [h@from-internal:1] Macro("SIP/200-00000007", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/200-00000007", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/200-00000007", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/200-00000007", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/200-00000007", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/200-00000007", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/200-00000007", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/200-00000007' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/200-00000007'
Scheduling destruction of SIP dialog 'MWUzNGZkNDE4OGU4YjdmYTg1ZTQ4YjA1ZTI3YTEwMGY.' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 192.168.1.14:50426 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.14:50426;branch=z9hG4bK-d8754z-b96c56174d467f7e-1---d8754z-;received=192.168.1.14;rport=50426
From: "200";tag=1b02126e
To: "300";tag=as1678386c
Call-ID: MWUzNGZkNDE4OGU4YjdmYTg1ZTQ4YjA1ZTI3YTEwMGY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>

<--- SIP read from 192.168.1.14:50426 --->
ACK sip:300@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.14:50426;branch=z9hG4bK-d8754z-b96c56174d467f7e-1---d8754z-;rport
To: "300";tag=as1678386c
From: "200";tag=1b02126e
Call-ID: MWUzNGZkNDE4OGU4YjdmYTg1ZTQ4YjA1ZTI3YTEwMGY.
CSeq: 2 ACK
Content-Length: 0

-------------------------------------------------------------------------------------------------

lo mas interesante de esto que les mostre anteriormente es lo siguiente..

Found user '200'
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found unknown media description format BV32 for ID 107
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.14:28326
Looking for 300 in from-internal (domain 192.168.1.5)
list_route: hop:

Problema de codecs ??????

bueno en fin despues como me hbla mucho del dialplan y medio pereza analizarlo todo.....lo siguiente que hice fue crear mi propio dial plan en extensions_custom.conf para poder analizar mejor

[interno]
exten => _xxx,1,Dial(SIP/${EXTEN},15,tT)
exten => _xxx,2,Voicemail(${EXTEN}@default)
exten => _xxx,3,playback(vm_goodbye)
exten => _xxx,4,hangup

Y RESULTA QUE HAGO LA LLAMADA Y SI ME DEJA HACERLA sin ningum problema....AHORA EL PROBLEMA RADICA EN EL DIALPLAN QUE ME GENERA FREEPBX ...

entonces he instalado ya 3 versiones distintas de freepbx y de asterisk y todavia tengo el mismo problema HABRA UNA PARAMETRO QUE NO HE MODIFICADO ?????' QUE YO RECUERDE EN ELASTIX CREABAS LAS EXTENSION Y LISTO LLAMABAS SIN NINGUM PROBLEMA .....

algun experto podria ayudarme

DISCULPEN POR OCUPAR PARTE DE SU TIEMPO ..GRACIAS

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