Sip trunk timeouts whatever I do

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Sip trunk timeouts whatever I do

Postby bagosm » Tue May 05, 2015 11:13 am

Hello, I keep trying to set up a SIP trunk properly but I always get timeout after 120 seconds. What I have working so far is DAHDI analog trunk (with outgoing and ingoing calls) and local extensions.

The same machine but with a different router used to properly connect to the same service. Now it's located in a LAN with 2 routers (192.168.1.x and 192.168.2.x) although it only uses the first one and ignores the 2nd one. Pinging from cli works fine to the required service's address (

Also, using the same information I connect properly to the SIP service from softphones on computers on the same network (192.168.1.x).

I have tried several configurations, including the official ones provided by and still nothing works, and I'm really lost as to how to debug this problem Any help I would appreciate much!

Also, here is some info (I have censored some values)

Code: Select all

Code: Select all
localhost*CLI> sip show settings

Global Settings:
  UDP Bindaddress:
  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-AsteriskNOW-12.0.62(11.16.0)
  SDP Session Name:       Asterisk PBX 11.16.0
  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:            On
  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:             external.ip.number.iuse:0
  Externrefresh:          10

Global Signalling Settings:
  Codecs:                 (g723|gsm|ulaw|alaw|g726|g729)
  Codec Order:            ulaw:20,alaw:20,gsm:20,g726:20,g729:20,g723:30
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          Yes
  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:            Yes
  DTMF:                   rfc2833
  Qualify:                0
  Keepalive:              0
  Use ClientCode:         No
  Progress inband:        Never
  Tone zone:              <Not set>
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   *97

Posts: 2
Joined: Tue May 05, 2015 11:06 am

Re: Sip trunk timeouts whatever I do

Postby david55 » Thu May 07, 2015 2:59 pm

Which version (your configuration uses option that deprecated, and may not work at all in supported versions)?

You will provide logs in sufficient detail, but I would assume a NAT or firewall problem
Moves Like Spencer
Posts: 12570
Joined: Fri Sep 26, 2008 5:03 am

Re: Sip trunk timeouts whatever I do

Postby bagosm » Wed May 13, 2015 8:46 am

Hello, thanks for the reply. I use the latest AsteriskNOW 6.12 32-bit version with FreePBX 12.0.63

Logs don't show much at all except for the constant:

[2015-05-13 17:44:23] NOTICE[2201] chan_sip.c: -- Registration for '' timed out, trying again (Attempt #46)

What else can I do to show more logs/what do you reffer to specifically because I am not experienced as you understand and I dont know any other place for logs!
Posts: 2
Joined: Tue May 05, 2015 11:06 am

Re: Sip trunk timeouts whatever I do

Postby david55 » Fri May 29, 2015 6:09 am

Enable the full log in logger.conf, then do sip set debug on.

The most likely cause is that you are behind NAT (you have no options that would allow the other side to find your public address.
Moves Like Spencer
Posts: 12570
Joined: Fri Sep 26, 2008 5:03 am

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