Summary: | ASTERISK-17034: [regression] Can't receive SIP INFO DTMF when using Read() without Answer() | ||
Reporter: | johnnylu (johnnylu) | Labels: | |
Date Opened: | 2010-11-26 23:20:39.000-0600 | Date Closed: | 2015-03-27 16:47:20 |
Priority: | Major | Regression? | Yes |
Status: | Closed/Complete | Components: | Channels/chan_sip/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | 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@192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK1055cc8709f8e9af From: <sip:135@192.168.1.101>;tag=5d478c2528f28482 To: <sip:511@192.168.1.102> User-Agent: Asterisk CSeq: 3 INFO Call-ID: 6eb1064116403e8852ec1d3c00080e9b@192.168.1.101 Contact: <sip:135@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@192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK2ed64d862aa64792 From: <sip:135@192.168.1.101>;tag=5d478c2528f28482 To: <sip:511@192.168.1.102> User-Agent: Asterisk CSeq: 4 INFO Call-ID: 6eb1064116403e8852ec1d3c00080e9b@192.168.1.101 Contact: <sip:135@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} | ||
Comments: | By: Matt Jordan (mjordan) 2015-03-27 16:47:05.274-0500 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. |