# 180 Ringing after 183 Progress is not passed on to the caller

**URL:** <https://community.asterisk.org/t/180-ringing-after-183-progress-is-not-passed-on-to-the-caller/91126>\
**Category:** Asterisk SIP\
**Created:** [January 5, 2022, 9:46am UTC](https://community.asterisk.org/t/180-ringing-after-183-progress-is-not-passed-on-to-the-caller/91126 "2022-01-05T09:46:47Z")\
**Posts on this page:** 7\
**Page:** 1

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**Author:** ![TheMark](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/themark/32/4771_2.png) [@TheMark](https://community.asterisk.org/u/TheMark)\
**Post date:** [January 5, 2022, 9:46am UTC](https://community.asterisk.org/t/180-ringing-after-183-progress-is-not-passed-on-to-the-caller/91126/1 "2022-01-05T09:46:47Z")

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Have a problem after upgrading from asterisk 1.8 to 18 with pjsip  
when a user make a outgoing call from there Asterisk PBX via our Asterisk GW to our provider  
the Asterisk GW newer indicate 180 Ringing to our Asterisk PBX  
the problem is our provider first send a 183 Progress with SDP and after that send a 180 Ringing  
with asterisk 1.8 this was not a problem as It would just send it, but with asterisk 18 will change it to 183

is this a bug or as intended ??

```auto
           178.21.248.59:5060 178.21.248.172:5060 195.215.224.12:5060
          ─;;;;;;;;;┬─;;;;;;;; ─;;;;;;;;;┬─;;;;;;;; ─;;;;;;;;;┬─;;;;;;;;
  10:44:06.916070 │ INVITE (SDP) │ │
        +0.000755 │ ─ > │ │
  10:44:06.916825 │ 100 Trying │ │
        +0.013465 │ <─<<<<<<<<<<<<<<<<<<<<<<<<< │ │
  10:44:06.930290 │ │ INVITE (SDP) │
        +0.202759 │ │ ─ > │
  10:44:07.133049 │ │ 100 Trying │
        +2.195055 │ │ <─<<<<<<<<<<<<<<<<<<<<<<<<< │
  10:44:09.328104 │ │ 183 Session Progress (SDP) │
        +0.001095 │ │ <─<<<<<<<<<<<<<<<<<<<<<<<<< │
  10:44:09.329199 │ │ PRACK │
        +0.000605 │ │ ─ > │
  10:44:09.329804 │ 183 Session Progress (SDP) │ │
        +0.055379 │ <─<<<<<<<<<<<<<<<<<<<<<<<<< │ │
  10:44:09.385183 │ │ 200 OK │
        +1.398832 │ │ <─<<<<<<<<<<<<<<<<<<<<<<<<< │
  10:44:10.784015 │ │ 180 Ringing │
        +0.000820 │ │ <─<<<<<<<<<<<<<<<<<<<<<<<<< │
  10:44:10.784835 │ │ PRACK │
        +0.000703 │ │ ─ > │
  10:44:10.785538 │ 183 Session Progress (SDP) │ │
        +0.045755 │ <─<<<<<<<<<<<<<<<<<<<<<<<<< │ │
  10:44:10.831293 │ │ 200 OK │
        +1.432871 │ │ <─<<<<<<<<<<<<<<<<<<<<<<<<< │
  10:44:12.264164 │ │ 200 OK │
        +0.000764 │ │ <─<<<<<<<<<<<<<<<<<<<<<<<<< │
  10:44:12.264928 │ │ ACK │
        +0.000641 │ │ ─ > │
  10:44:12.265569 │ 200 OK (SDP) │ │
        +0.014951 │ <─<<<<<<<<<<<<<<<<<<<<<<<<< │ │
  10:44:12.280520 │ ACK │ │
        +2.936439 │ ─ > │ │
  10:44:15.216959 │ │ BYE │
        +0.000885 │ │ <─<<<<<<<<<<<<<<<<<<<<<<<<< │
  10:44:15.217844 │ │ 200 OK │
        +0.002238 │ │ ─ > │
  10:44:15.220082 │ BYE │ │
        +0.010167 │ <─<<<<<<<<<<<<<<<<<<<<<<<<< │ │
  10:44:15.230249 │ 200 OK │ │

```

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [January 5, 2022, 11:51am UTC](https://community.asterisk.org/t/180-ringing-after-183-progress-is-not-passed-on-to-the-caller/91126/2 "2022-01-05T11:51:17Z")

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According to the code[1] and issue[2] it is on purpose.

[1] [asterisk/chan\_pjsip.c at master · asterisk/asterisk · GitHub](https://github.com/asterisk/asterisk/blob/master/channels/chan_pjsip.c#L1636)  
[2] [[ASTERISK-27994] PJSIP: Early media ringback not indicated after Progress() - Digium/Asterisk JIRA](https://issues.asterisk.org/jira/browse/ASTERISK-27994)

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<div class="post-metadata">

**Author:** ![TheMark](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/themark/32/4771_2.png) [@TheMark](https://community.asterisk.org/u/TheMark)\
**Post date:** [January 6, 2022, 4:48pm UTC](https://community.asterisk.org/t/180-ringing-after-183-progress-is-not-passed-on-to-the-caller/91126/3 "2022-01-06T16:48:49Z")

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I have looked at the link you send and from what I understand there problem was that Asterisk did not send INBAND ringing for the Dial application if the channel already had early media set up via Progress  
not where asterisk send 180 or 183

the problem is that only 180 Ringing will be send out on AMI  
so changing 180 to 183 prevent the caller from setting the correct AST\_STATE\_RINGING

do you see any problem with this patch suggestion

```auto
chan_pjsip.c: prevent changing 180 to 183 if early_media already enabled

index e8fbb3d79a..882db3cd6c 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -1638,11 +1638,9 @@ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const voi
                        if (channel->session->endpoint->inband_progress ||
                                (channel->session->inv_session && channel->session->inv_session->neg &&
                                pjmedia_sdp_neg_get_state(channel->session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE)) {
- response_code = 183;
                                res = -1;
- } else {
- response_code = 180;
                        }
+ response_code = 180;
                } else {
                        res = -1;
                }

```

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [January 6, 2022, 4:50pm UTC](https://community.asterisk.org/t/180-ringing-after-183-progress-is-not-passed-on-to-the-caller/91126/4 "2022-01-06T16:50:58Z")

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I don’t know. I didn’t make the original change, and that was some time ago so can’t recall any further specifics.

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<div class="post-metadata">

**Author:** ![TheMark](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/themark/32/4771_2.png) [@TheMark](https://community.asterisk.org/u/TheMark)\
**Post date:** [January 6, 2022, 4:54pm UTC](https://community.asterisk.org/t/180-ringing-after-183-progress-is-not-passed-on-to-the-caller/91126/5 "2022-01-06T16:54:41Z")

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ok, thanks

I will try man a jira ticket and submit my patch and add as much details at possible and see where it lands

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<div class="post-metadata">

**Author:** ![TheMark](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/themark/32/4771_2.png) [@TheMark](https://community.asterisk.org/u/TheMark)\
**Post date:** [January 6, 2022, 7:45pm UTC](https://community.asterisk.org/t/180-ringing-after-183-progress-is-not-passed-on-to-the-caller/91126/6 "2022-01-06T19:45:14Z")

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[https://issues.asterisk.org/jira/browse/ASTERISK-29842](https://issues.asterisk.org/jira/browse/ASTERISK-29842)

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<div class="post-metadata">

**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\
**Post date:** [February 5, 2022, 7:45pm UTC](https://community.asterisk.org/t/180-ringing-after-183-progress-is-not-passed-on-to-the-caller/91126/7 "2022-02-05T19:45:17Z")

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Closed on February 5, 2022, 7:45pm UTC

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