Hola Buenos dias,
Soy nuevo con asterisk y con todo lo que conlleva esto, el asunto es que estoy instalando mi central telefonica, esta en un servidor en la nube, configure mi troncal con los siguientes datos
nombre de la troncal: troncal_miami
detalles del par
host=sip1.3nx.co
username=3052606246
secret=XXXXXXXXXXXXX
type=friend
port=5060
Para las opciones de entrada no coloque nada, segun vi en algunos otros foros.
configure mi ruta de llamadas salientes, sin inconveniente (puedo realizar llamadas sin problema)
mi problema es con las llamadas entrantes, quiero que cuando alguien llame al 3052606246 me desvie la llamada a la extension que tengo configurada en mi softphone (utilizo zoiper, version gratuita)
aqui esta la salida del CLI cuando realizo la llamada a mi DID number
-- Executing [13052606246@from-sip-external:1] NoOp("SIP/46.19.209.14-00000102", "Received incoming SIP connection from unknown peer to 13052606246") in new stack
-- Executing [13052606246@from-sip-external:2] Set("SIP/46.19.209.14-00000102", "DID=13052606246") in new stack
-- Executing [13052606246@from-sip-external:3] Goto("SIP/46.19.209.14-00000102", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/46.19.209.14-00000102", "0?checklang:noanonymous") in new stack
-- Goto (from-sip-external,s,5)
-- Executing [s@from-sip-external:5] Set("SIP/46.19.209.14-00000102", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2019-09-26 07:39:15.522 -05.
-- Executing [s@from-sip-external:6] Log("SIP/46.19.209.14-00000102", "WARNING,"Rejecting unknown SIP connection from 46.19.209.14"") in new stack
[2019-09-26 07:39:00] WARNING[10718][C-000000f2]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from 46.19.209.14"
-- Executing [s@from-sip-external:7] Answer("SIP/46.19.209.14-00000102", "") in new stack
> 0x7f21c0051370 -- Strict RTP switching to RTP remote address 46.19.209.71:48686 as source
-- Executing [s@from-sip-external:8] Wait("SIP/46.19.209.14-00000102", "2") in new stack
[2019-09-26 07:39:00] NOTICE[10718][C-000000f2]: channel.c:4304 __ast_read: Dropping incompatible voice frame on SIP/46.19.209.14-00000102 of format alaw since our native format has changed to (ulaw)
> 0x7f21c0051370 -- Strict RTP learning complete - Locking on source address 46.19.209.71:48686
-- Executing [s@from-sip-external:9] Playback("SIP/46.19.209.14-00000102", "ss-noservice") in new stack
-- <SIP/46.19.209.14-00000102> Playing 'ss-noservice.gsm' (language 'en')
-- Executing [s@from-sip-external:10] PlayTones("SIP/46.19.209.14-00000102", "congestion") in new stack
-- Executing [s@from-sip-external:11] Congestion("SIP/46.19.209.14-00000102", "5") in new stack
== Spawn extension (from-sip-external, s, 11) exited non-zero on 'SIP/46.19.209.14-00000102'
-- Executing [h@from-sip-external:1] Hangup("SIP/46.19.209.14-00000102", "") in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/46.19.209.14-00000102'
agradesco de antemano, sus respuestas... Gracias
