e-fon (CH) und Asterisk (Codec Probleme?)

[maikd]

Hi zusammen,

Ich hab meinen Asterisk nun endlich aufgesetzt und alles scheint zu funktionieren. Leider bekomme ich bei e-fon (CH) immer beim raustelefonieren (rein geht) die folgenden Meldungen auf der Konsole:

-- Executing Dial("SIP/100-1860", "SIP/044857xxxx@efon|60|tr") in new stack
-- Called 044857xxxx@efon
-- SIP/efon-7a9f is making progress passing it to SIP/100-1860
Sep 26 15:58:35 NOTICE[-178238544]: channel.c:1724 ast_set_read_format: Unable to find a path from G729A to ULAW
Sep 26 15:58:35 NOTICE[-178238544]: channel.c:1691 ast_set_write_format: Unable to find a path from ULAW to G729A
Sep 26 15:58:35 WARNING[-178238544]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (20)?
Sep 26 15:58:35 WARNING[-178238544]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (20)?
Sep 26 15:58:35 WARNING[-178238544]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (20)?

usw...

Sep 26 15:58:37 WARNING[-178238544]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (20)?
Sep 26 15:58:37 WARNING[-178238544]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (20)?
-- SIP/efon-7a9f answered SIP/100-1860
-- Attempting native bridge of SIP/100-1860 and SIP/efon-7a9f
Sep 26 15:58:37 NOTICE[-178238544]: channel.c:1724 ast_set_read_format: Unable to find a path from G729A to ULAW
Sep 26 15:58:37 NOTICE[-178238544]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G729A
Sep 26 15:58:37 WARNING[-178238544]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (20)?
Sep 26 15:58:37 WARNING[-178238544]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/100-1860(4) to SIP/efon-7a9f(256)
Sep 26 15:58:37 WARNING[-178238544]: channel.c:2633 ast_channel_bridge: Can't make SIP/100-1860 and SIP/efon-7a9f compatible
Sep 26 15:58:37 WARNING[-178238544]: res_features.c:358 ast_bridge_call: Bridge failed on channels SIP/100-1860 and SIP/efon-7a9f
== Spawn extension (ext100, 044857xxxx, 1) exited non-zero on 'SIP/100-1860'
-- Executing Hangup("SIP/100-1860", "") in new stack
== Spawn extension (ext100, h, 1) exited non-zero on 'SIP/100-1860'


Hat das auch schon mal wer gehabt?

 

[wesco]

Bluewin Phone

Hallo, habe unten in deinem eintrag bei "Provider" Bluewin Phone gesehen. Hast du Diesen Account schon mit Asterisk zum laufen gebracht?? bei mir geht nur raus, ankommende anrufe bekommen nach ca. 30s ein besetzt.
Grüsse aus Schaffhausen....