SIP: Timeout on non-critical invite transaction.

VitalPBX Community Support General Discussion SIP: Timeout on non-critical invite transaction.

  • Post

    Hello ,

    I am keep getting

    [2019-04-22 07:44:38] WARNING[6171]: chan_sip.c:4178 retrans_pkt: Timeout on 254985259-1270243963-1187409528 on non-critical invite transaction.

    My Sip Show Settings are

    Global Settings:
    UDP Bindaddress:
    TCP SIP Bindaddress: Disabled
    TLS SIP Bindaddress: Disabled
    RTP Bindaddress: Disabled
    Videosupport: No
    Textsupport: No
    Ignore SDP sess. ver.: No
    AutoCreate Peer: Off
    Match Auth Username: No
    Allow unknown access: No
    Allow subscriptions: Yes
    Allow overlap dialing: Yes
    Allow promisc. redir: No
    Enable call counters: Yes
    SIP domain support: No
    Path 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: VitalPBX
    SDP Session Name: Asterisk PBX 16.2.1
    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: asterisk
    From: Domain:
    Record SIP history: 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: AF41
    802.1p CoS SIP: 3
    802.1p CoS RTP audio: 5
    802.1p CoS RTP video: 4
    802.1p CoS RTP text: 3
    Jitterbuffer enabled: No

    Network Settings:
    SIP address remapping: Enabled using externaddr
    Externhost: <none>
    Externrefresh: 10

    Global Signalling Settings:
    Codecs: (ulaw|alaw|g729)
    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:No
    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
    Max forwards: 70

    Default Settings:
    Allowed transports: UDP
    Outbound transport: UDP
    Context: sip-default
    Record on feature: one_touch_
    Record off feature: one_touch_
    Force rport: Yes
    DTMF: rfc2833
    Qualify: 0
    Keepalive: 0
    Use ClientCode: No
    Progress inband: No
    Language: en
    Tone zone: us
    MOH Interpret: default
    MOH Suggest:
    Voice Mail Extension: *97
    RTCP Multiplexing: No

    I am using Twilio and Trunk – Sometime when I call my DID it fail and sometime it connect, but at first I am sure it’s something to do with above warning, How can I fix this. Thanks

Viewing 7 replies - 1 through 7 (of 7 total)
  • Replies

    That’s just a warning

    Do you have any issue? What’s your real problem?


    About this message: “chan_sip.c: Timeout on 356135904-147483397-409121765 on non-critical invite transaction.”

    This may happen for the following reasons:

    • A NAT device in the signaling path. A misconfigured NAT device is in the signaling path and stops SIP messages.
    • A firewall that blocks messages or reroutes them wrongly in an attempt to assist in a too clever way.
    • A SIP middlebox (SBC) that rewrites contact: headers so that we can’t reach the other side with our reply or the ACK.
    • A badly configured SIP proxy that forgets to add record-route headers to make sure that signaling works.
    • Packet loss. IP and UDP are unreliable transports. If you lose too many packets the retransmits don’t help and communication is impossible. If this happens with signaling, media would be unusable anyway.
    • Some routers are configured by default with the “SIP ALG” option. In some cases, this option should be off because the implementation is incomplete.
    • Unauthorized invites. Also, you may get this message when SIP scanners try to vulnerate your server.

    Similar as to what I’m experiencing. Are your phones going ‘unreachable’??

    This just started happening to me on my hosted phone instance.



    You should try to connect a softphone or a desktop phone from your home/office network to discard a bad configuration on the PBX side.



    My softphone from desktop and cellphone (zoiper) connects just fine…  so perhaps it is customer location related and I’m in disbelieve.




    As I told you in the other post, I think this is related to your customer network configuration, maybe some kind of firewall policy that is blocking the invites send from the PBX to his network.

    You may try to use OpenVPN to connect one of your customer’s phones for proving that is something wrong on his network. Just make sure the OpenVPN port is open on the customer and PBX side.


    I’ll keep testing. And report back.  Just baffled, that is all.  As I manage the firewall and network for this location/customer and nothing has changed.

    I will go onsite this evening. Could be another vendor was there and made change or tried to put some equipment inline that I’m not seeing.

Viewing 7 replies - 1 through 7 (of 7 total)


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