Receiving sporadic 503 Service Unavailable

We are getting a CONGESTION after answering the call. It is only happening with one service provider and not every time. I’m curious to know how the SIP signaling works.

We are using asterisk ARI. It’s an inbound scenario where we dial a party-B number.

  • Answer Inbound call

  • Dial party-B

  • Create a bridge on party-B answer

  • Add party-A and party-B channels to bridge

  • Record bridge

And we are getting CONGESTION after RecordingStarted event. Is there any SIP request exchange happening after call answer (200 OK)? while adding to bridge or recording

{ type: 'Dial',
  dialstring: '',
  timestamp: '2022-03-03T10:41:36.926+0530',
  forward: '',
> dialstatus: 'CONGESTION',
   { id: 'd0a8e0e4-0f34-4991-9b56-4128cb53672b',
    > state: 'Up',
     name: 'SIP/vod-00006902',
     caller: { name: '', number: 'XXXXXXXXX' },
     language: 'en',
     connected: { name: '', number: 'XXXXXXXXXX' },
     accountcode: '',
     dialplan: { context: 'incoming', exten: '', priority: 1 },
     creationtime: '2022-03-03T10:41:29.478+0530' },
   asterisk_id: '78:ac:c0:fe:85:42',
   application: 'XXXXX' }

Yes there can be SIP transactions after the OK. However, they can’t result in a CONGESTION status.

Why not just look at the SIP logging

Please note that chan_sip is deprecated, unsupported, and scheduled for removal.

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Thanks for the reply. Sure we will look into the SIP log for more details.

Still can you tell me some possible events after OK.

Thanks for the chan_sip warning. We are planning to update Asterisk and SIP within couple of months. what’s your say on running asterisk 16 over 18 on production? currently using 14.7

Version support timelines are documented on the wiki[1]. Asterisk 16 goes security fix only October 9th of this year.

[1] Asterisk Versions - Asterisk Project - Asterisk Project Wiki

Thanks @jcolp . Sure I will check that.

Can you share something more about the SIP events after OK?

I couldn’t find anything even after searching for long time. That’s why commented here and this is my first interaction in this community and didn’t expected a fast response. Once again thank you guys

Without an actual SIP log, there’s nothing to really be said.

Re-INVITE to change media destination e.g. direct media.
Re-INVITE to hold or unhold a media stream.
Re-INVITE to add or delete a media stream (not (well) supported by Asterisk).
Re-INVITE to update connected line information.
Re-INVITE to reset session timers.
UPDATE for at least some of these.

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Thanks @david551.

We found the issue, actually it was a SIP 500 error not 503. And stasis log showed as a dial congestion.

We found a INVITE after the call answer and and it caused an 500 error at network operator end. I think it was due to the missing canreinvite=no configuration in that particular SIP account. We updated the SIP configuration, and didn’t find congestion in any answered channel in last couple of hours. Still monitoring.

Once again thanks for your support.

I assume you meant directmedia=no, as canreinvite has been deprecated for about a decade. Under either name, this only affects:

canreinvite=no is working as I expected in asterisk 14.7 with chan_sip. I added it because, it was already there in other SIP accounts and they are working fine.

I read it somewhere before. Will try the directmedia instead of canreinvite

Thank you

They are synonyms, but canreinvite was deprecated, because it only blocks one type of re-INVITE.

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