[Asterisk-code-review] channels/chan sip: 180 Ringing not sent after 183 Session Pr... (asterisk[13])

[Morten Tryfoss]

Morten Tryfoss has posted comments on this change.

Change subject: channels/chan_sip: 180 Ringing not sent after 183 Session Progress
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Patch Set 1:

We're interconnecting with an Ericsson MSC/AXE. There are some issues between SIP and ISUP when the doesn't enter RINGING state, even though early media is active. This switch sends 183 always first, and 180 when it recieves CPG (ISUP) from the network.

Another problem is that for example a "480 Temporarily Unavailable" is interpreted differently if a call is ringing or not. I gives release cause 20 instead of 19 if it has not entered RINGING (which gives our customers a false message stating that "the cell phone is outside coverage").

If UA's got problem with this, they should probably use progressinband=yes?

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Gerrit-MessageType: comment
Gerrit-Change-Id: I498fed853128831b80536f8818cd7b60e641f39c
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Morten Tryfoss <morten at tryfoss.no>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
Gerrit-Reviewer: Morten Tryfoss <morten at tryfoss.no>
Gerrit-HasComments: No