Сисетма FreeBSD 6.1 + Asterisk 1.4.4 собран из портов, карта Wildcard X100P
При исходящем звонке на zap интерфейс отличная слышимость с двух сторон, а при входящем вызове треск во время слов от звонящего абонента. При тишине треска нет. То есть ситуация такая: я с sip отвечаю на вызов и меня отлично слышно а я слышу с треском, помехи как будто по радио телефону.
городская линия отличная и шумов в ней нет

Вот конфиг:
extensions.conf:------------------------------
[incoming]
exten => s,1,wait(0)
exten => s,2,Answer()
exten => s,3,Dial(SIP/3035)
[pstn-dialout]
exten => s,1,Wait, 1
exten => s,2, Answer
exten => s,3, BackGround(hello-world)
exten => s,4, Read(ext)
exten => s,5, Dial(ZAP/g1/${ext:1},20,rT)
----------------------------------------------
zapata.conf:-----------------------------------
[trunkgroups]
[channels]
language=en
context=incoming
signalling=fxs_ks
cidsignalling=dtmf
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
callreturn=yes
rxgain=-4.0
txgain=5.0
group=1
busydetect=yes
busycount=3
callgroup=1
callprogress=no
progzone=us
pickupgroup=1
immediate=yes
musiconhold=default
channel => 1
--------------------------------------------
zaptel.conf:---------------------------------
fxsks=1
loadzone = us
defaultzone=us
--------------------------------------------

[root@webby ~]# ztcfg -vv
Keyword: [fxsks], Value: [1]
Keyword: [loadzone], Value: [us]
Keyword: [defaultzone], Value: [us]

Zaptel Version: 1.4.0-BSD
Echo Canceller: MG2
Configuration
======================


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels configured.

[root@webby ~]#

Логи при исходящем звонке:
-- Executing [3500@default:1] Goto("SIP/3035-0873b000", "pstn-dialout|s|1") in new stack
-- Goto (pstn-dialout,s,1)
-- Executing [s@pstn-dialout:1] Wait("SIP/3035-0873b000", " 1") in new stack
-- Executing [s@pstn-dialout:2] Answer("SIP/3035-0873b000", "") in new stack
-- Executing [s@pstn-dialout:3] BackGround("SIP/3035-0873b000", "hello-world") in new stack
-- <SIP/3035-0873b000> Playing 'hello-world' (language 'en')
[Jul 28 12:31:36] NOTICE[28787]: rtp.c:783 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. C
lient IP: 10.10.10.24
-- Executing [s@pstn-dialout:4] Read("SIP/3035-0873b000", "ext") in new stack
-- User entered '77212206'
-- Executing [s@pstn-dialout:5] Dial("SIP/3035-0873b000", "ZAP/g1/7212206|20|rT") in new stack
-- Called g1/7212206
-- Zap/1-1 answered SIP/3035-0873b000
-- Hungup 'Zap/1-1'
== Spawn extension (pstn-dialout, s, 5) exited non-zero on 'SIP/3035-0873b000'
-- Executing [5555@default:1] Goto("SIP/3035-0873b000", "voicemenu-custom-1|s|1") in new stack
-- Goto (voicemenu-custom-1,s,1)
-- Executing [s@voicemenu-custom-1:1] Answer("SIP/3035-0873b000", "") in new stack
== Auto fallthrough, channel 'SIP/3035-0873b000' status is 'UNKNOWN'

Логи при входящем звонке:
- Starting simple switch on 'Zap/1-1'
[Jul 28 12:44:38] NOTICE[28787]: chan_zap.c:6365 ss_thread: Got event 18 (Ring Begin)...
[Jul 28 12:44:38] NOTICE[28787]: chan_zap.c:6365 ss_thread: Got event 2 (Ring/Answered)...
[Jul 28 12:44:42] NOTICE[28787]: chan_zap.c:6365 ss_thread: Got event 18 (Ring Begin)...
-- Executing [s@incoming:1] Wait("Zap/1-1", "0") in new stack
-- Executing [s@incoming:2] Answer("Zap/1-1", "") in new stack
-- Executing [s@incoming:3] Dial("Zap/1-1", "SIP/3035") in new stack
-- Called 3035
-- SIP/3035-0873b000 is ringing
-- Call on SIP/3035-0873b000 left from hold
-- SIP/3035-0873b000 answered Zap/1-1
[Jul 28 12:44:50] NOTICE[28787]: rtp.c:783 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. C
lient IP: 10.10.10.24
== Spawn extension (incoming, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'

Кто знает что это такое?
я и подваление эхо включал и выключал - все равно та же проблема - треск

формат текста неудачно получился

Сисетма FreeBSD 6.1 + Asterisk 1.4.4 собран из портов, карта Wildcard X100P При исходящем звонке на zap интерфейс отличная слышимость с двух сторон, а при входящем вызове треск во время слов от звонящего абонента. При тишине треска нет. То есть ситуация такая: я с sip отвечаю на вызов и меня отлично слышно а я слышу с треском, помехи как будто по радио телефону. городская линия отличная и шумов в ней нет <

Вот конфиг:
extensions.conf:------------------------------
[incoming]
exten => s,1,wait(0)
exten => s,2,Answer()
exten => s,3,Dial(SIP/3035)
[pstn-dialout]
exten => s,1,Wait,1
exten => s,2, Answer
exten => s,3, BackGround(hello-world)
exten => s,4, Read(ext)
exten => s,5, Dial(ZAP/g1/${ext:1},20,rT)
----------------------------------------------
zapata.conf:-----------------------------------
[trunkgroups]
[channels]
language=en
context=incoming
signalling=fxs_ks
cidsignalling=dtmf
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
callreturn=yes
rxgain=-4.0
txgain=5.0
group=1
busydetect=yes
busycount=3
callgroup=1
callprogress=no
progzone=us
pickupgroup=1
immediate=yes
musiconhold=default
channel => 1
--------------------------------------------
zaptel.conf:--------------------------------
fxsks=1
loadzone = us
defaultzone=us
--------------------------------------------
[root@webby ~]# ztcfg -vv
Keyword: [fxsks], Value: [1]
Keyword: [loadzone], Value: [us]
Keyword: [defaultzone], Value: [us]
Zaptel Version: 1.4.0-BSD
Echo Canceller: MG2 Configuration
======================
Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
[root@webby ~]#

Логи при исходящем звонке:
-- Executing [3500@default:1] Goto("SIP/3035-0873b000", "pstn-dialout|s|1") in new stack
-- Goto (pstn-dialout,s,1)
-- Executing [s@pstn-dialout:1] Wait("SIP/3035-0873b000", " 1") in new stack
-- Executing [s@pstn-dialout:2] Answer("SIP/3035-0873b000", "") in new stack
-- Executing [s@pstn-dialout:3] BackGround("SIP/3035-0873b000", "hello-world") in new stack
-- <SIP/3035-0873b000> Playing 'hello-world' (language 'en')
[Jul 28 14:04:55] NOTICE[29147]: rtp.c:783 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. C
lient IP: 10.10.10.24
-- Executing [s@pstn-dialout:4] Read("SIP/3035-0873b000", "ext") in new stack
-- User entered '77891120'
-- Executing [s@pstn-dialout:5] Dial("SIP/3035-0873b000", "ZAP/g1/7891120|20|rT") in new stack
-- Called g1/7891120
-- Zap/1-1 answered SIP/3035-0873b000
-- Hungup 'Zap/1-1'
== Spawn extension (pstn-dialout, s, 5) exited non-zero on 'SIP/3035-0873b000'

Логи при входящем звонке:
-- Starting simple switch on 'Zap/1-1'
[Jul 28 14:07:11] NOTICE[29147]: chan_zap.c:6365 ss_thread: Got event 18 (Ring Begin)...
[Jul 28 14:07:12] NOTICE[29147]: chan_zap.c:6365 ss_thread: Got event 2 (Ring/Answered)...
[Jul 28 14:07:15] NOTICE[29147]: chan_zap.c:6365 ss_thread: Got event 18 (Ring Begin)...
-- Executing [s@incoming:1] Wait("Zap/1-1", "0") in new stack
-- Executing [s@incoming:2] Answer("Zap/1-1", "") in new stack
-- Executing [s@incoming:3] Dial("Zap/1-1", "SIP/3035") in new stack
-- Called 3035
-- SIP/3035-08743000 is ringing
-- Call on SIP/3035-08743000 left from hold
-- SIP/3035-08743000 answered Zap/1-1
[Jul 28 14:07:22] NOTICE[29147]: rtp.c:783 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. C
lient IP: 10.10.10.24
== Spawn extension (incoming, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'


Кто знает что это такое? я и подваление эхо включал и выключал - все равно та же проблема - треск