| Summary: | ASTERISK-11089: POTS->VoIP calling defficiency | ||
| Reporter: | Maxim Sobolev (sobomax) | Labels: | |
| Date Opened: | 2007-12-20 04:40:31.000-0600 | Date Closed: | 2008-01-15 15:15:47.000-0600 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_zap |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) chan_zap.c | |
| Description: | There is a problem with call that comes via analogue Zaptel interface and then is hanged up before it has been answered there is no way to communicate rejection to the POTS side (i.e. you can't hang up before it has been aswered). Therefore when the next ring pulse comes in shortly on the same POTS call it's considered to be a "new" call and so that the new call is initiated. This continues until the calling party on the POTS side hangs up. I have attached very crude patch that we have had for years to help with this situation by waiting until the calling party in turn hangs up (by detecting ring timeout). -Maxim | ||
| Comments: | By: Russell Bryant (russell) 2007-12-20 15:51:38.000-0600 What hardware are you using that has this problem? By: Maxim Sobolev (sobomax) 2007-12-20 16:18:06.000-0600 Any analogue PSTN hardware will have this problem. TDM400P is the one in our case. -Maxim By: Jason Parker (jparker) 2008-01-15 15:15:47.000-0600 Unfortunately, this isn't something that can be fixed, and that's due to issues with analog signaling itself, rather than something inside of Asterisk. The easiest way around this would be to answer the incoming call before sending it on to the SIP device, but that would cause a CDR to be created, ringing would be stopped, and your provider would likely charge you for the call. You could also try to do some dialplan magic, and keep the incoming side of the call up after the Dial(), and check DIALSTATUS, then do an answer and immediate hangup. | ||