DTMF from "Send DTMF" button fails after upgrade
Posted: Fri Mar 07, 2014 4:09 pm
VERSION: 2.8-424a
BUILD: 140126-2253
He Guys,
Apparently, after upgrading from 2.4 to 2.6 to 2.8 (in batches short after each other) and coming from asterisk 1.2 and now running asterisk 1.4.44-vici, we just found out DTMF isn't working the way it was anymore. We use an IAX2 trunk, which is just the same as before.
The symptoms are as follows:
1) agent logs in, paused;
2) agent places manual call, i.e. to DTMF test number 31857732598 (speaks out loud every DTMF tone you send);
3) agent then sends DTMF code "1" via "Send DTMF" button;
4) Asterisk CLI tells agi-dtmf is called:
5) But nothing happens. A channel is opened and hungup after playing '1'. It is as if the tones aren't being sent over the line.
When I use the connected soft or hard phone and use that one to send out DTMF tones, it all works fine (but indeed, we surpass the MeetMe room and agi-dtmf script then. Though the script seems to be running fine (all the "silence" playbacks are for every character in "unknown" what the calleridname is being passed to agi-dtmf.agi, for whatever reason.
Any clue what might be missing here?
Regards,
Kim
BUILD: 140126-2253
He Guys,
Apparently, after upgrading from 2.4 to 2.6 to 2.8 (in batches short after each other) and coming from asterisk 1.2 and now running asterisk 1.4.44-vici, we just found out DTMF isn't working the way it was anymore. We use an IAX2 trunk, which is just the same as before.
The symptoms are as follows:
1) agent logs in, paused;
2) agent places manual call, i.e. to DTMF test number 31857732598 (speaks out loud every DTMF tone you send);
3) agent then sends DTMF code "1" via "Send DTMF" button;
4) Asterisk CLI tells agi-dtmf is called:
- Code: Select all
[Mar 7 22:00:04] -- Executing [8500998@default:3] AGI("Local/8500998@default-05da,2", "agi-dtmf.agi") in new stack
[Mar 7 22:00:04] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-dtmf.agi
[Mar 7 22:00:04] -- Playing 'silence' (escape_digits=) (sample_offset 0)
[Mar 7 22:00:04] -- Playing '1' (escape_digits=) (sample_offset 0)
[Mar 7 22:00:04] -- Playing 'silence' (escape_digits=) (sample_offset 0)
[Mar 7 22:00:05] -- Playing 'silence' (escape_digits=) (sample_offset 0)
[Mar 7 22:00:05] -- Playing 'silence' (escape_digits=) (sample_offset 0)
[Mar 7 22:00:05] -- Playing 'silence' (escape_digits=) (sample_offset 0)
[Mar 7 22:00:05] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 7 22:00:05] -- Playing 'silence' (escape_digits=) (sample_offset 0)
[Mar 7 22:00:06] -- Playing 'silence' (escape_digits=) (sample_offset 0)
[Mar 7 22:00:06] -- Playing 'silence' (escape_digits=) (sample_offset 0)
[Mar 7 22:00:06] -- Playing 'silence' (escape_digits=) (sample_offset 0)
[Mar 7 22:00:06] -- Playing 'silence' (escape_digits=) (sample_offset 0)
[Mar 7 22:00:07] == Parsing '/etc/asterisk/manager.conf': [Mar 7 22:00:07] Found
[Mar 7 22:00:07] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 7 22:00:07] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 7 22:00:07] -- AGI Script agi-dtmf.agi completed, returning 0
[Mar 7 22:00:07] -- Executing [8500998@default:4] Hangup("Local/8500998@default-05da,2", "") in new stack
[Mar 7 22:00:07] == Spawn extension (default, 8500998, 4) exited non-zero on 'Local/8500998@default-05da,2'
[Mar 7 22:00:07] -- Executing [h@default:1] DeadAGI("Local/8500998@default-05da,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
5) But nothing happens. A channel is opened and hungup after playing '1'. It is as if the tones aren't being sent over the line.
When I use the connected soft or hard phone and use that one to send out DTMF tones, it all works fine (but indeed, we surpass the MeetMe room and agi-dtmf script then. Though the script seems to be running fine (all the "silence" playbacks are for every character in "unknown" what the calleridname is being passed to agi-dtmf.agi, for whatever reason.
Any clue what might be missing here?
Regards,
Kim