Summary:ASTERISK-15420: ISDN to SIP doesn't generate SIP 180 Ringing with Call Progress ISDN message
Reporter:Denis Galvao (denisgalvao)Labels:
Date Opened:2010-01-13 10:57:17.000-0600Date Closed:2011-06-07 14:08:11
Versions:Frequency of
Description:When I place a call trough a SIP UA and this call goes out through an ISDN link, the Call Progress message from ISDN is not converted to a SIP 180 Ringing message to the UA.

P.S.: Im not sure about the category of this issue, BTW I believe it is a channel related thing.


I believe that this is a problem related to Asterisk because this IETF document tells that 180 ringing should appear on this ISDN x SIP scenario.

This is the reference(Section 2.2 page 15):

This is my system:
E1 Card: Digium TE412P - 3rd gen
Dahdi: DAHDI Version: 2.2.

Comments:By: Birger "WIMPy" Harzenetter (wimpy) 2010-01-13 23:39:18.000-0600

I think the quoted document is getting it wrong.

A proceeding message only means the setup was acceptred for switching, but not that there is a remote party ringing. That will be indicated by an alterting message.
A proceeding message should translate to 183 Trying or something.

At this point it might be worth noting that a call can be connected, without having been in the ringing state.
Not only the proceeding, but also the alerting message are optional.

By: Denis Galvao (denisgalvao) 2010-01-14 07:19:44.000-0600

Wimpy: I accept your thoughts, but what to do when no Alerting messages come from the ISDN leg? Is there a way reproduce that behaviour?

By: Birger "WIMPy" Harzenetter (wimpy) 2010-01-14 10:07:13.000-0600

I'm not sure what the actual problem is.
I would translate the messages as described above.

And what do you want to reproduce? A connect without alerting?
That happens where calls are auto-answered, so will almost certainly be true for data calls. IVR systems might also do it. However there's no way to do it with Asterisk AFAIK even if that might suggest itself for calls answered within the dialplan.

By: Denis Galvao (denisgalvao) 2010-01-14 11:59:59.000-0600

What I have in my scenario is:
Some placed calls from SIP that goes through an ISDN link doesn't send to my SIP ua the 180 ringing, because there is no Alerting coming from ISDN. What I need is to always generate a ringing, with or without the alerting message(except when a BUSY message come).

By: Birger "WIMPy" Harzenetter (wimpy) 2010-01-14 12:07:00.000-0600

I'm not a SIP expert. Does SIP require a Ringing message before a connect?
In that case the only solution I see is a state machine that sends a ringing when the connect message is received and no Ringing was sent so far.

Translating a proceeding message into ringing will surely cause strange effects.

By: Denis Galvao (denisgalvao) 2010-01-15 12:12:25.000-0600

No, 1xx messages are not required in a SIP call flow.

By: Marcos Jose Setim (msetim) 2010-01-18 05:41:37.000-0600


In the meantime, I ask myself why five authors A. Johnston, S. Donovan, R. Sparks, C. Cunningham, K. Summers accepted that flow. I don't know in deep about ISDN and SIP interworking however, the document "contains best current practice examples of Session  Initiation Protocol (SIP) call flows showing interworking with the Public Switched Telephone Network (PSTN)" (PAGE 1).

I'm not saying that either wimpy or denisgalvao are wrong (your arguments really make sense) but I'm bring a new perspective trying to understand why they wrote that flow. I don't found a RFC talking about interworking between SIP/ISDN.

By: Birger "WIMPy" Harzenetter (wimpy) 2010-01-18 14:45:59.000-0600

I haven't got a clue as to why someone would want to translate the ISDN messages to anything else than their SIP equivalents. Atfter all they translate very well.
So I don't think Asterisk is doing anything wrong here. It looks perfectely reasonable to me.

What is it you're calling that's answering without ringing and why do you need ringing?
Would option r to dial be a solution to your problem?

By: Denis Galvao (denisgalvao) 2010-01-19 06:26:54.000-0600

I(actually not me) have a business rule on my application that analyze all 180 Ringing messages to do an action based on that SIP message

Ok, this could be changed, I thnk the same, but the person behind this application doesn't like the idea, so I start to learn about this internetworking call flows and found that doc.

I don't want to change the Asterisk behaviour based on my needs, but based on the best practices
Could we have some thoughts from an Asterisk developer here?

By: Leif Madsen (lmadsen) 2010-01-29 16:15:08.000-0600

Can you please provide the console output, with debugging enabled, along with the SIP console trace which describes this issue?

We'll need the information in order to track down why this isn't working because this should already be working.

By: Leif Madsen (lmadsen) 2010-01-29 16:15:47.000-0600

Could you also provide a PRI debug as well? Please attach as files to this issue. Thanks!

By: Leif Madsen (lmadsen) 2010-01-29 16:16:57.000-0600

Also, if you could provide a description of what is happening, and what you're expecting to happen, that would also be useful in determining what we should do here.

By: Leif Madsen (lmadsen) 2010-02-03 14:04:51.000-0600

I'm closing this issue as after having a brief discussion with Russell, he thinks this is expected behaviour, and is an issue in the reporters application.