wrong number trouble

forb
Joined: Thu 21 of Jan, 2010

wrong number trouble

Posted:Thu 21 of Jan, 2010 (09:30 UTC)
Hello, there are a bunch of ubuntu server + asterisk 1.6.2 + freepbx 2.6.0 + Account sipnet.ru

Incoming calls go, but with outgoing have trouble.
If the call through the trunk sipnet.ru to a wrong number - i get an error 503, and after this error is impossible to call the existing number (drop the same error 503), it is necessary to wait about 2 minutes or re-register again.

Global Settings:

UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: Disabled
TLS SIP Port: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: No
Direct RTP setup: No
User Agent: Asterisk PBX 1.6.2.0
SDP Session Name: Asterisk PBX 1.6.2.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms

Network QoS Settings:

IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No

Network Settings:

SIP address remapping: Enabled using externip
Externhost: <none>
Externip: 91.193.цык.цык:5060
Externrefresh: 10
Internal IP: 10.10.0.3:5060
Localnet: 10.10.0.0/255.255.255.0
STUN server: 0.0.0.0:0

Global Signalling Settings:

Codecs: 0xe (gsm|ulaw|alaw)
Codec Order: ulaw:20,alaw:20,gsm:20
Relax DTMF: No
RFC2833 Compensation: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Nat: Always
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97