| Summary: | ASTERISK-00200: DTMF and early audio via chan_sip blocks Asterisk if set to inband | ||
| Reporter: | paul_cheng (paul_cheng) | Labels: | |
| Date Opened: | 2003-08-31 17:51:03 | Date Closed: | 2004-09-25 02:46:18 |
| Priority: | Blocker | Regression? | No |
| Status: | Closed/Complete | Components: | Core/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | This bug may be related to the other reports of * not responding to reload, restart now, etc. and artifact SIP channels. There are actually two bugs to report, so I will submit this report twice with each one separated by title. When dialing a number which reaches an IVR system (usually toll-free numbers) via channel SIP and audio is received before the call is "answered", DTMF will not pass correctly. If the call is "answered" then DTMF passes correctly. If DTMFMODE=rfc2833 or DTMFMODE=info, then the IVR just keeps going and does not detect DTMF. However, if DTMFMODE=inband and user presses a number, then the call goes blank and Asterisk does not hangup the call. Input at the console is then ignored for restart, reload, etc. however, sip show peers, etc. still work. The only way to stop asterisk is to "kill" from a shell command line. EXAMPLE: Using a SIP UA, dial 18008828880 (or 18004354000) via FWD or any other SIP Proxy which lets you connect an 800 call. NOTE: Not all SIP UAs will accept early audio. Cisco ATAs will. Grandstreams will not. Try DTMF using rfc2833 or info or inband. If it is set to inband, asterisk will block and not release the channel. ****** ADDITIONAL INFORMATION ****** LATEST CVS as of 2 minutes ago RH9 | ||
| Comments: | By: Mark Spencer (markster) 2003-09-03 23:28:58 This is going to require ssh access to your box to debug. By: pcheng (pcheng) 2003-09-04 15:30:03 Will you need root access or just a user with admin privileges? We have different boxes that you can access. Early audio is something we still haven't really gotten a handle on. We have two boxes both running Redhat 9 with the same kernel, packages, etc. and we copied the Asterisk source and config files from one box to the other. With Cisco ATAs, early audio works on both boxes, but with other devices like Addpac Voicefinders, the early audio works on one box, but not the other. We still haven't figured it out. In any case, that's a different issue. Let me know if you need root access and I'll send username and pw privately. By: Mark Spencer (markster) 2003-09-23 13:47:22 I will need root access. Find me on IRC (irc.freenode.net, #asterisk, kram) By: Mark Spencer (markster) 2003-09-24 14:46:49 I placed another fix in CVS this morning. Can you confirm if *this* fixes it? By: tekp (tekp) 2003-09-26 17:07:28 I had the exact same issues as detailed in the bug description above. I use inband dtmf and asterisk would hang approx. every couple hours the same way paul mentioned, only if I tried to upgrade asterisk to any version after CVS-07/16/2003, well that's the date I built it, I don't remember when I downloaded it. Maybe a cpl days earlier. The version I have running from 09/24/03 has been up now for 9 hours with relatively high call volume. So this issue seems fixed for me. By: Mark Spencer (markster) 2003-09-26 19:37:03 Fixed in CVS | ||