How to connect Asterisk and iPECS-MG via SIP trunk?

C

CompBOO2014-10-27 15:22:24

Asterisk

CompBOO, 2014-10-27 15:22:24

Good afternoon.
I'm trying to set up a bunch of Asterisk and iPECS-MG via a SIP trunk, Asterisk acts as a child server. The main server is ipecs-mg.
I set up a trunk on an asterisk, set up reception on LG, the asterisk logs show that the registration is going through, but when I call, it says that it will not connect.
PBXs are on the same network, there is no NAT. If to call on the contrary, then silence and in logs of the Asterisk is pure.

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3 answer(s)

V

Vladimir, 2014-10-27
@rostel

exhaust:

asterisk -rx "sip show settings"
asterisk -rx "sip show peers"
asterisk -rx "sip show registry"

Show

C

CompBOO, 2014-10-28
@CompBOO

localhost*CLI> sip show settings


Global Settings:
----------------
  UDP Bindaddress:        0.0.0.0:5060
  TCP SIP Bindaddress:    Disabled
  TLS SIP Bindaddress:    Disabled
  Videosupport:           No
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        Off
  Match Auth Username:    No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Realm. auth:            No
  Our auth realm          asterisk
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Direct RTP setup:       No
  User Agent:             FPBX-2.11.0(11.12.0)
  SDP Session Name:       Asterisk PBX 11.12.0
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Trust RPID:             No
  Send RPID:              No
  Legacy userfield parse: No
  Send Diversion:         Yes
  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:          4294967295
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

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

Network Settings:
---------------------------
  SIP address remapping:  Enabled using externaddr
  Externhost:             <none>
  Externaddr:             172.18.2.106:0
  Externrefresh:          10
  Localnet:               172.18.0.0/255.255.0.0
                          10.118.0.0/255.255.0.0

Global Signalling Settings:
---------------------------
  Codecs:                 (alaw)
  Codec Order:            alaw:20
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          No
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            30
  RTP Hold Timeout:       300
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         No
  Pedantic SIP support:   Yes
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Sub. min duration       60 secs
  Sub. max duration:      3600 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Outbound reg. retry 403:0
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      Yes
  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
  Max forwards:           70

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:     UDP
  Context:                from-sip-external
  Record on feature:      automon
  Record off feature:     automon
  Force rport:            No
  DTMF:                   rfc2833
  Qualify:                0
  Keepalive:              0
  Use ClientCode:         No
  Progress inband:        Never
  Language:
  Tone zone:              <Not set>
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   *97

sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
600/600                   172.18.20.96                             D  Yes        Yes         A  5060     OK (4 ms)
ipecs                     172.18.2.8                                  Yes        Yes            5060     OK (16 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]

sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
0 SIP registrations.

O

olegsmrnff, 2015-06-23
@olegsmrnff

LG IPECS LIK 50 SIP trunk Asterisk
asterisk-pbx.ru/wiki/hardware/pbx/lg_ipecs