Summary: | ASTERISK-10642: progressinband=yes send 180 and 183 together | ||
Reporter: | kfsoo01 (kfsoo01) | Labels: | |
Date Opened: | 2007-10-29 15:45:44 | Date Closed: | 2007-11-06 01:52:47.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | I set progressinband=yes on asterisk-1.4.13. Asterisk sent 180 Ringing and 183 Session Progress together, which causing the ringback tone broken. Asterisk had sent 180 ringing first. Asterisk sent 183 when got inband progress indication from pstn. ****** ADDITIONAL INFORMATION ****** <--- Transmitting (NAT) to 10.11.162.244:53540 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.11.162.244:5060;received=10.11.162.244 From: <sip:10.11.162.244>;tag=271A8798-7B1 To: <sip:1781@10.11.161.237>;tag=as25d40ea6 Call-ID: FD603E3D-859511DC-BAB7A1F1-BDB9CA89@10.11.162.244 CSeq: 101 INVITE User-Agent: astpbx01 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:1781@10.11.161.237> Content-Length: 0 <------------> Audio is at 10.11.161.237 port 16288 Adding codec 0x100 (g729) to SDP <--- Transmitting (NAT) to 10.11.162.244:53540 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.11.162.244:5060;received=10.11.162.244 From: <sip:10.11.162.244>;tag=271A8798-7B1 To: <sip:1781@10.11.161.237>;tag=as25d40ea6 Call-ID: FD603E3D-859511DC-BAB7A1F1-BDB9CA89@10.11.162.244 CSeq: 101 INVITE User-Agent: astpbx01 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:1781@10.11.161.237> Content-Type: application/sdp Content-Length: 213 | ||
Comments: | By: kfsoo01 (kfsoo01) 2007-10-29 15:46:47 Can Asterisk don't send 180 and 183 together when progressinband=yes?? By: Joshua C. Colp (jcolp) 2007-10-31 19:48:11 It's not possible to get rid of the 180 Ringing as you seem to need, but it is perfectly acceptable to send 180 and then 183. The SIP implementation should just accept the audio stream from the 183 Session Progress and stop producing it's own ringing. By: Olle Johansson (oej) 2007-11-06 01:52:47.000-0600 There's no error in sending a 183 after sending a 180. Thanks for reporting what you suspected was a bug though. If it sounds strange on the end device, that's something for that software to fix. |