есть Asterisk 1,4

файлик sip.conf

[general]
context=corp
allowguest=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
pedantic=yes
videosupport=yes
relaxdtmf=yes
dtmfmode=auto

useclientcode=yes
language=ru

register => XXX:XXX@sipnet.ru

[sipnet.ru]
type=friend
username=XXXXXXXX
secret=XXXXXXXXXXXXXX
fromuser=XXXXXXXXXXXX
fromdomain=sipnet.ru
host=sipnet.ru
port=5060
canreinvite=no
nat=yes
dtmfmode=auto
insecure=very
;reinvite=no
;context=hg
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
allow=gsm

[Cisco-1] (cisco с карточкой E1 смотрит в атс)
username=asterisk
type=friend
secret=12345-sip
qualify=yes
nat=yes
insecure=very
context=corp
host=172.**.**.**
disallow=all
allow=g729
allow=alaw
allow=ulaw

[Solomenka-1] Удалённай цыска с поднятым кал-менеджером
username=asterisk
type=friend
secret=12345
qualify=yes
nat=yes
;insecure=invite
insecure=very
context=corp
host=193.**.**.**
disallow=all
allow=g729
allow=ulaw
allow=alaw


[6555] (Ip Phone 7940)
username=6555
type=friend
secret=12345
qualify=yes
nat=yes
host=dynamic
caninvite=no
context=corp

[6250] (Cisco ATA-188)
type=friend
secret=12345-sip
qualify=yes
nat=yes
canreinvite=no
host=dynamic
caninvite=no
canreinvite=no
context=corp



Конфиг цыски

version 12.4
service nagle
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
boot-start-marker
boot system flash
boot-end-marker
!
logging buffered 10000 debugging
aaa new-model
!
!
aaa session-id common
!
resource policy
!
clock timezone EST 2
clock summer-time UKR-SUMMER recurring last Sun Mar 2:00 last Sun Oct 2:00
network-clock-participate wic 0
!
!
ip cef
!
!
isdn switch-type primary-net5
isdn voice-call-failure 0
isdn gateway-max-interworking
!
voice-card 0
codec complexity high
dspfarm
no local-bypass
!
!
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
h323
sip
registrar server expires max 3600 min 3600
redirect contact order best-match
no call service stop
!
!
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g726r16
codec preference 3 g729r8
!
!
!
!
!
!
!
!
!
!
!
!
voice translation-rule 1
rule 1 /^1/ /611/ type any unknown plan isdn unknown
rule 2 /^2/ /612/ type any unknown plan isdn unknown
rule 3 /^2/ /613/ type any unknown plan isdn unknown
!
voice translation-rule 2
rule 1 /../ // type subscriber unknown plan isdn unknown
!
!
voice translation-profile TEST
translate called 1
!
voice translation-profile TEST-in-POTS
translate calling 2
!
!
!
!
call-history-mib retain-timer 0
call-history-mib max-size 500
!
!
controller E1 0/0/0
framing NO-CRC4
pri-group timeslots 1-16
!
!
!
!
!
!
interface GigabitEthernet0/0
ip address 172.16.129.77 255.255.254.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
!
interface Serial0/0/0:15
no ip address
encapsulation hdlc
no logging event link-status
dialer idle-timeout 2147483
isdn switch-type primary-net5
isdn protocol-emulate network
isdn incoming-voice modem
isdn sending-complete
no cdp enable
no clns route-cache
!
ip route 0.0.0.0 0.0.0.0 172.***.***.***
!
!
no ip http server
no ip http secure-server
!
!
control-plane
!
voice-port 0/0/0:15
translation-profile incoming TEST-in-POTS
cptone RU
bearer-cap 3100Hz
!
!
!
!
dial-peer cor custom
!
!
!
dial-peer voice 100 pots
destination-pattern 1..
direct-inward-dial
port 0/0/0:15
prefix 1
!
dial-peer voice 200 pots
destination-pattern 2..
direct-inward-dial
port 0/0/0:15
prefix 2
!
dial-peer voice 300 pots
destination-pattern 3..
direct-inward-dial
port 0/0/0:15
prefix 3
!
dial-peer voice 6 voip
destination-pattern 6...
voice-class codec 1
session protocol sipv2
session target ipv4:172.16.128.70
dtmf-relay rtp-nte
fax rate 9600
fax protocol pass-through g711alaw
no vad
!
dial-peer voice 400 pots
destination-pattern 0T
direct-inward-dial
port 0/0/0:15
!
!
gateway
timer receive-rtp 1200
!
sip-ua
authentication username ******* password ************
nat symmetric check-media-src
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
mwi-server ipv4:172.16.128.70 expires 3600 port 5060 transport udp
registrar ipv4:172.16.128.70 expires 3600
sip-server ipv4:172.16.128.70

sh version
c2800nm-adventerprisek9_ivs-mz.124-9.T.bin

трабл возник из неоткуда всё работало себе спокойненько до определённого момента.
потом звонки с АТА-шек дохотят но в разных вариантах ктото когото не слышит а гдето всё нормально.
Удалённых включений около сотни.

Каналы на удалёнки криптованы темеже cisco.

Много буков.

Ситуация исследуемая, инструменатрий известный - дебаг на цисках и на истерисках.

Цыска пол беды. Между аташкими тоже траблы.

w*CLI>
<--- SIP read from 172.16.132.252:5060 --->
REGISTER sip:172.16.128.70 SIP/2.0
Via: SIP/2.0/UDP 172.16.132.252:5060;branch=z9hG4bK7f2584f54d1d3209
From: <sip:6132@172.16.128.70;user=phone>;tag=824179359
To: <sip:6132@172.16.128.70;user=phone>
Call-ID: 2628145614@172.16.132.252
CSeq: 184796 REGISTER
Contact: <sip:6132@172.16.132.252:5060;user=phone;transport=udp>;expires=120
User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A)
Authorization: Digest username="6132",realm="asterisk",nonce="4e274c46",uri="sip:172.16.128.70",response="39fc68989d726283ea430c151dcbfbd4"
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 172.16.132.252 : 5060 (no NAT)
w*CLI>
<--- Transmitting (NAT) to 172.16.132.252:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.132.252:5060;branch=z9hG4bK7f2584f54d1d3209;received=172.16.132.252
From: <sip:6132@172.16.128.70;user=phone>;tag=824179359
To: <sip:6132@172.16.128.70;user=phone>
Call-ID: 2628145614@172.16.132.252
CSeq: 184796 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6132@172.16.128.70>
Content-Length: 0

маршрутизация между 172.16.132.252 и 172.16.128.70, фильтры iptables, и аксцесс листы.
Судя по
nat=yes
у вас юзеры за НАТом? Это зачем НАТ между 172.16.132.252 и 172.16.128.70?

canreinvite=no ; это понятно
host=dynamic
caninvite=no ; а это нет. Где Вы такое видели?

далее, если есть
sip-ua
authentication username ******* password ************
mwi-server ipv4:172.16.128.70 expires 3600 port 5060 transport udp
registrar ipv4:172.16.128.70 expires 3600
sip-server ipv4:172.16.128.70

то
dial-peer voice 6 voip
session target sip-server
а не
ipv4:
в раздел Бизнес, подраздел Работа - удалённо - требуется.

Нет ната нету. Фильтров нету.
nat=yes потому что в противном случае не работает.

ded:

маршрутизация между 172.16.132.252 и 172.16.128.70

traceroute 172.16.132.252
traceroute to 172.16.132.252 (172.16.132.252), 64 hops max, 40 byte packets
1 gw-1 (172.16.129.51) 0.828 ms 0.566 ms 0.863 ms
2 core1 (172.33.1.2) 1.735 ms 1.081 ms 0.852 ms
3 192.168.128.10 (192.168.128.10) 19.976 ms 34.824 ms 41.192 ms
4 172.16.132.252 (172.16.132.252) 42.205 ms 77.549 ms 58.817 ms

со стороны 172.16.128.70
permit ip 172.16.128.0 0.0.1.255 172.16.132.0 0.0.0.255
deny ip any any

со стороны 172.16.132.252
permit ip 172.16.132.0 0.0.0.255 172.16.128.0 0.0.1.255
deny ip any any

маршруты статические с обеих сторон.