App_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

General Help
1

Dear Experts,

I have FreePbx 2.11 with accessone trunk

From time to time I have problem with dialing out and noticed a lot of these:
WARNING[4028][C-00005d05] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)

This is just warning and not error so it should affect my trunk - am I right?

FreePbx is behind firewall with NAT enabled and external IP. Trunk is set as simple as possible:

host=IP Provider
type=peer
context=from-trunk-sip-accessone

user details:
type=peer
host=IP Provider

I have tried many different settings with no luck.

2
  • Name : accessone
    Description :
    Secret :
    MD5Secret :
    Remote Secret:
    Context : from-trunk-sip-accessone
    Record On feature : automon
    Record Off feature : automon
    Subscr.Cont. :
    Language :
    Tonezone :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Named Callgr :
    Nam. Pickupgr:
    MOH Suggest :
    Mailbox :
    VM Extension : *97
    LastMsgsSent : 0/0
    Call limit : 0
    Max forwards : 0
    Dynamic : No
    Callerid : “” <>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : no
    Force rport : Yes
    Symmetric RTP: Yes
    ACL : No
    DirectMedACL : No
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: 4294967295
    DirectMedia : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Ign SDP ver : No
    Trust RPID : No
    Send RPID : No
    TrustIDOutbnd: Legacy
    Subscriptions: Yes
    Overlap dial : Yes
    DTMFmode : rfc2833
    Timer T1 : 500
    Timer B : 32000
    ToHost : IP
    Addr->IP : IP:5060
    Defaddr->IP : (null)
    Prim.Transp. : UDP
    Allowed.Trsp : UDP
    Def. Username:
    SIP Options : (none)
    Codecs : (gsm|ulaw|alaw)
    Codec Order : (gsm:20,alaw:20,ulaw:20)
    Auto-Framing : No
    Status : Unmonitored
    Useragent :
    Reg. Contact :
    Qualify Freq : 60000 ms
    Keepalive : 0 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    RTP Engine : asterisk
    Parkinglot :
    Use Reason : No
    Encryption : No
3

Your reference:-

http://networking.ringofsaturn.com/Routers/isdncausecodes.php

specifically

Cause No. 20 - subscriber absent.
This cause value is used when a mobile station has logged off. Radio contact is not obtained with a mobile station or if a personal telecommunication user is temporarily not addressable at any user-network interface.

for a more diagnostic view , “sip set debug ip (IP Provider)[:port]” and for resolution, ensure your freeepbx_routes/routers/firewall/network/VSP/CID/Codecs are all adequate and in agreement for the call to “get through”

4

Thank You dicko,

I should have known “sip set debug ip”. I will let know my finding when this is resolved.

-MST