SPA3102 linea x DSL

[rickygm]

Hola foro , tengo un comportamiento algo extraño con spa3102 , si yo lo pongo a trabajar con una linea análoga pura, recibo y hago llamadas perfectamente , pero si la linea sale a través de un DSL , recibo llamadas perfectamente , pero cuando hago una me sale esto en el cli de asterisk:
-- Executing [922686894@chinamedsa:1] Dial("SIP/103-0000000c", "SIP/22686894@ata3102,40,rRtT") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/22686894@ata3102
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [922686894@chinamedsa:2] Congestion("SIP/103-0000000c", "") in new stack
== Spawn extension (chinamedsa, 922686894, 2) exited non-zero on 'SIP/103-0000000c'

alguna idea?

[rickygm]

yo recibo una llamada perfecto :
-- Executing [s@from-tele:1] NoOp("SIP/ata3102-0000001a", "") in new stack
-- Executing [s@from-tele:2] Goto("SIP/ata3102-0000001a", "IVRMAIN,s,1") in new stack
-- Goto (IVRMAIN,s,1)
-- Executing [s@IVRMAIN:1] Wait("SIP/ata3102-0000001a", "1") in new stack
-- Executing [s@IVRMAIN:2] Set("SIP/ata3102-0000001a", "CHANNEL(language)=es") in new stack
-- Executing [s@IVRMAIN:3] Set("SIP/ata3102-0000001a", "TIMEOUT(digit)=4") in new stack
-- Digit timeout set to 4.000
-- Executing [s@IVRMAIN:4] Set("SIP/ata3102-0000001a", "TIMEOUT(response)=5") in new stack
-- Response timeout set to 5.000
-- Executing [s@IVRMAIN:5] Wait("SIP/ata3102-0000001a", "1") in new stack
-- Executing [s@IVRMAIN:6] BackGround("SIP/ata3102-0000001a", "ext-or-zero") in new stack
> 0x7612ecd8 -- Probation passed - setting RTP source address to 192.168.1.102:16212
-- <SIP/ata3102-0000001a> Playing 'ext-or-zero.alaw' (language 'es')
-- Executing [s@IVRMAIN:7] WaitExten("SIP/ata3102-0000001a", "8") in new stack
-- Timeout on SIP/ata3102-0000001a, going to 't'
-- Executing [t@IVRMAIN:1] SIPAddHeader("SIP/ata3102-0000001a", "Alert-Info:Office") in new stack
-- Executing [t@IVRMAIN:2] Dial("SIP/ata3102-0000001a", "SIP/102&SIP/103,22,t") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/102
-- Called SIP/103
-- SIP/103-0000001c connected line has changed. Saving it until answer for SIP/ata3102-0000001a
-- SIP/102-0000001b connected line has changed. Saving it until answer for SIP/ata3102-0000001a
-- SIP/103-0000001c is ringing
-- SIP/103-0000001c is ringing
-- SIP/102-0000001b is ringing
-- SIP/103-0000001c is ringing
-- SIP/103-0000001c is ringing
-- SIP/103-0000001c connected line has changed. Saving it until answer for SIP/ata3102-0000001a
-- SIP/103-0000001c answered SIP/ata3102-0000001a
> 0x76626940 -- Probation passed - setting RTP source address to 192.168.1.17:5048
== Spawn extension (IVRMAIN, t, 2) exited non-zero on 'SIP/ata3102-0000001a'
pbx*CLI>

[rickygm]

anexo una imagen de la config de mi ata

[rickygm]

anexo una imagen de la config de mi ata

[rickygm]

hola foro , alguna idea , este tema lo tengo con una ?

[Christian Cabrera]

Audiblemente hay diferencia entre una línea u otra?

Me suena a que con la linea DSL el ATA no detecta el tono o el voltaje de la línea, por lo que no procede. Quizá tenga que ver con el tema de las cadencias configuradas en el ATA.

[rickygm]

Audiblemente hay diferencia entre una línea u otra?

Me suena a que con la linea DSL el ATA no detecta el tono o el voltaje de la línea, por lo que no procede. Quizá tenga que ver con el tema de las cadencias configuradas en el ATA.

Hola Christian , No realmente , la voz normal , cuando recibo no tengo problemas , pero para sacar la llamada hay no mas la corta , yo estoy en nicaragua y no he visto nada para este pais , que tipo de cadencia optar .

en base a tu experiencia que me recomiendas.

[hector]

Trata de sacar una grabación del dial tone y analízalo con Audacity para que verifiques que la configuración es correcta

[rickygm]

Hola, después de googlear y revisar el tema del voltaje de la linea este me ayudo :

International Control Note:When dialing out if you get a SIP 503 error ([CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]) when dialing out check the Voice->info tab for 'Line Voltage'. I have seen reports where it is as high as 48V, but in my case it was 22V idle, and 6V-7V in use. I got the in-use voltage by calling in then checking the info page again. It defaults to 30V, so I assume it thought the line was always in use (though dial-in still worked) so it would give the 503 error. You may or may not need to change this but changing this to 10V fixed my error.

sacado de https://wiki.freeswitch.org/wiki/SPA...eeSwitch_HowTo

realmente cuando llamaba mi voltaje era de 7V y en iddle era 55V , lo setee a 10V y funciona.

gracias por la ayuda.