180 Ringing Missing

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Asterisk Users 7 Comments

hi,

after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know, its old. customer is very conservative…)

i have problem with missing 180 Ringing

flow is easy (PBX -> Asterisk -> SIP SBC)

Asterisk 11
PBX – Asterisk
-> INVITE
<- 100 Trying <- 183 Session Progress ( <- RTP -> )
<- 180 Ringing <- 200 OK Asterisk 13 -> INVITE
<- 100 Trying <- 183 Session Progress ( <- RTP -> )

__MISSING RINGING___

<- 200 OK temporarily i solved problem with using "R" param R: Default: Indicate ringing to the calling party, even if the called party     isn't actually ringing. Allow interruption of the ringback if early media     is received on the channel. it changed to Asterisk 13 (Dial(${ARG1},300,R) -> INVITE
<- 100 Trying <- 180 Ringing <- 183 Session Progress ( <- RTP -> )
<- 200 OK any ideas why Ringing is missing? any solutions? Marek --

7 thoughts on - 180 Ringing Missing

  • Have you compared the signaling in both directions between the two versions to see if there is a difference?

  • If you have 183 Session progress, there is no need to send 180 Ringing (especially not AFTER 183 Session progress), as you already have early media instead. Having both is actually a bit misleading IMHO.

    So this is actually correct. One should not rely on any of these 1xx “Provisional” messages. They may or may not be sent, without violating SIP standards.

    Am 01.12.20, 12:20 schrieb “asterisk-users im Auftrag von marek” :

    hi,

    after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know,
    its old. customer is very conservative…)

    i have problem with missing 180 Ringing

    flow is easy (PBX -> Asterisk -> SIP SBC)

    Asterisk 11
    PBX – Asterisk
    -> INVITE
    <- 100 Trying <- 183 Session Progress ( <- RTP -> )
    <- 180 Ringing <- 200 OK Asterisk 13 -> INVITE
    <- 100 Trying <- 183 Session Progress ( <- RTP -> )

    __MISSING RINGING___

    <- 200 OK temporarily i solved problem with using “R” param R: Default: Indicate ringing to the calling party, even if the called party isn’t actually ringing. Allow interruption of the ringback if early media is received on the channel. it changed to Asterisk 13 (Dial(${ARG1},300,R) -> INVITE
    <- 100 Trying <- 180 Ringing <- 183 Session Progress ( <- RTP -> )
    <- 200 OK any ideas why Ringing is missing? any solutions? Marek —

  • Dne 01/12/2020 v 12:58 Joshua C. Colp napsal(a):
    whats your goal with this question?

    asking if there are some side effects in incoming call ? (SBC ->
    Asterisk -> PBX)

  • No side effects, but looking at the actual SIP signaling (sip set debug on) and see what the remote side is sending for SIP responses as well.

  • I disagree (and I think rfc3261 agrees). How would the caller know if the callee has been alerted if he doesn’t receive the 180?

    183 and 180 have different meaning. 180 indicates the callee is being alerted. An 183 has no such meaning but is often used to setup early media (although any 1xx can do that). 180 and 183 are not mutually exclusive and in fact form a full matrix of possible and useful states. There can be early media without any phone ringing (announcement), ringing without early media (caller is expected to generate a ringback tone itself), no early media nor ringing and finally early media and ringing simultaneous.

    Especially automata calling will want to know the difference.

    Of course, a generated ringback tone by the caller should be stopped when media is received.

    Asterisk should indicate to the caller the same state it received from the callee.
    —
    Alex Hermann

    —

  • Dne 01/12/2020 v 19:18 Alex Hermann napsal(a):

    for the record

    tried PJSIP only configuration and its the same. no 180 ringing

    (2.2.2.2 is PBX, 1.1.1.1 is Asterisk13+PJSIP connected to PSTN through SIP SBC)

      1 2020-12-18 17:12:28.656569 2.2.2.2 -> 1.1.1.1 SIP/SDP 976 Request:
    INVITE sip:222333444@1.1.1.1:5080 | , with session description
      2 2020-12-18 17:12:28.657113 1.1.1.1 -> 2.2.2.2 SIP 393 Status: 100
    Trying |
      3 2020-12-18 17:12:30.102754 1.1.1.1 -> 2.2.2.2 SIP/SDP 863 Status:
    183 Session Progress | , with session description
      4 2020-12-18 17:12:30.102828 1.1.1.1 -> 2.2.2.2 SIP/SDP 863 Status:
    183 Session Progress | , with session description
      5 2020-12-18 17:12:30.125256 1.1.1.1 -> 2.2.2.2 SIP/SDP 863 Status:
    183 Session Progress | , with session description
      6 2020-12-18 17:12:30.125385 1.1.1.1 -> 2.2.2.2 SIP/SDP 863 Status:
    183 Session Progress | , with session description
      7 2020-12-18 17:12:30.395564 1.1.1.1 -> 2.2.2.2 SIP/SDP 863 Status:
    183 Session Progress | , with session description
      8 2020-12-18 17:12:35.765423 1.1.1.1 -> 2.2.2.2 SIP/SDP 897 Status:
    200 OK | , with session description
      9 2020-12-18 17:12:35.767759 2.2.2.2 -> 1.1.1.1 SIP 480 Request: ACK
    sip:1.1.1.1:5080 |
     10 2020-12-18 17:12:37.891798 1.1.1.1 -> 2.2.2.2 SIP 506 Request: BYE
    sip:777888999@2.2.2.2:5060 |
     11 2020-12-18 17:12:37.894507 2.2.2.2 -> 1.1.1.1 SIP 566 Status: 200 OK |

    interesting note: on SIP SBC side i see 180 Ringing from Telco Operator

    i’ll try find if there is some recommendation –
    https://www.sipforum.org/technology/sipconnect/

    Marek

    —