Configuracion de troncal sip

[Jairo]

tengo un problema que supuestamente ya me registre con mi proveedor SIP pero no me dejan sacar ni entrar llamadas, ya cree un dialplan de salida en el extensions.conf pero me da este error.. sip is circuit-busy

[Christian Cabrera]

Necesitarias poner el sip debug que te arroja el proveedor cuando envías la llamada para entender que es lo que está causando la negación de la misma.

[Jairo]

Este error me tira al querer hacer una llamada

[Apr 10 09:54:58] WARNING[1460]: chan_sip.c:4174 retrans_pkt: Retransmission tim
eout reached on transmission 20efa1525164739675c751b54041a615@imshwi.claro.com.g
t for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/displ...etransmissions
Packet timed out after 6399ms with no response
[Apr 10 09:54:58] WARNING[1460]: chan_sip.c:4203 retrans_pkt: Hanging up call 20
efa1525164739675c751b54041a615@imshwi.claro.com.gt - no reply to our critical pa
cket (see https://wiki.asterisk.org/wiki/displ...etransmissions).
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [41503222@micontexto:3] Hangup("SIP/usuario1-00000057", "") in new stack
== Spawn extension (micontexto, 41503222, 3) exited non-zero on 'SIP/usuario1-00000057'
Scheduling destruction of SIP dialog 'OWI2NmZhNTk4YTM1MzQ3NWViNTdjZTFlZTdlMDQyOTI.' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 192.168.0.128:5060 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.0.128:5060;branch=z9hG4bK-d8754z-6a7cd8522909889d-1---d
8754z-;received=192.168.0.128
From: "Jairo"<sip:usuario1@192.168.0.125;transport=UDP>; tag=3508e819
To: <sip:41503222@192.168.0.125;transport=UDP>;tag=as2 509883d
Call-ID: OWI2NmZhNTk4YTM1MzQ3NWViNTdjZTFlZTdlMDQyOTI.
CSeq: 2 INVITE
Server: Asterisk PBX 11.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[Jairo]

cuando intento sacar una llamada:

== Using SIP RTP CoS mark 5
-- Executing [41503222@micontexto:1] Set("SIP/usuario1-00000073", "CALLERID(all)="+50222568920<sip:+50222568920@imsh wi.claro.com.gt>") in new stack
-- Executing [41503222@micontexto:2] Dial("SIP/usuario1-00000073", "SIP/carrier-claro/41503222,30,tr") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/carrier-claro/41503222
-- Got SIP response 502 "Bad Gateway" back from iphost:5060
-- SIP/carrier-claro-00000074 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
[Apr 10 10:34:38] WARNING[2910][C-00000049]: pbx.c:6456 __ast_pbx_run: Timeout,but no rule 't' or 'e' in context 'micontexto'
localhost*CLI>