[asterisk-bugs] [JIRA] (ASTERISK-17034) [regression] Can't receive SIP INFO DTMF when using Read() without Answer()

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-17034?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225673#comment-225673 ] 

Matt Jordan commented on ASTERISK-17034:
----------------------------------------

I don't think this is a bug.

In your log snippet, the {{INFO}} request does not contain a to-tag. As such, Asterisk would be unable to match the {{INFO}} request to a dialog, and would simply drop it. This would have worked prior to 1.8 as Asterisk ran with {{pedantic=no}} by default. Setting that back to {{no}} in 1.8 would probably mask the issue.

The {{INFO}} request may not have the to-tag for a variety of reasons:
# Your endpoint may be faulty.
# By indicating Ringing prior to Progress, you may be inhibiting sending of a 183.
# You may have {{progressinband}} enabled.

With the lack of a full DEBUG log on the issue along with a corresponding {{sip.conf}}, it is impossible to say for certain. However, this doesn't look like a bug.

> [regression] Can't receive SIP INFO DTMF when using Read() without Answer()
> ---------------------------------------------------------------------------
>
>                 Key: ASTERISK-17034
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-17034
>             Project: Asterisk
>          Issue Type: Bug
>          Components: Channels/chan_sip/General
>            Reporter: johnnylu
>
> I use Read command to read dtmf like:
> {noformat}
> Read(digito,input,20,n,3,30)
> {noformat}
> It's works good in asterisk 1.6.2.7, but it's not work when I update to asterisk 1.8.1-rc1
> when I change to {{Read(digito,input,20,i,3,30)}} it's works ok, but I  want't answer the line.
> My Exten:
> {noformat}
> exten =>_X.,1,Ringing()
> exten =>_X.,2,Progress()
> exten =>_X.,3,Playback(welcome)
> exten =>_X.,4,Read(digito,input,20,n,3,30)
> {noformat}
> and i revice the sip info:
> {noformat}
> <------------->
> --- (12 headers 2 lines) ---
> <--- SIP read from UDP:192.168.1.101:5060 --->
> INFO sip:511 at 192.168.1.102 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK1055cc8709f8e9af
> From: <sip:135 at 192.168.1.101>;tag=5d478c2528f28482
> To: <sip:511 at 192.168.1.102>
> User-Agent: Asterisk
> CSeq: 3 INFO
> Call-ID: 6eb1064116403e8852ec1d3c00080e9b at 192.168.1.101
> Contact: <sip:135 at 192.168.1.101:5060>
> Max-Forwards: 70
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
> Content-Type: application/dtmf-relay
> Content-Length: 22
> Signal=8
> Duration=0
> <------------->
> --- (12 headers 2 lines) ---
> <--- SIP read from UDP:192.168.1.101:5060 --->
> INFO sip:511 at 192.168.1.102 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK2ed64d862aa64792
> From: <sip:135 at 192.168.1.101>;tag=5d478c2528f28482
> To: <sip:511 at 192.168.1.102>
> User-Agent: Asterisk
> CSeq: 4 INFO
> Call-ID: 6eb1064116403e8852ec1d3c00080e9b at 192.168.1.101
> Contact: <sip:135 at 192.168.1.102:5060>
> Max-Forwards: 70
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
> Content-Type: application/dtmf-relay
> Content-Length: 22
> Signal=8
> Duration=0
> <------------->
> {noformat}



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