Asterisk: calling to external bridge

Asterisk Asterisk Support
1

Hi,

I have asterisk 1.2.24 and working fine, when i dial to an external bridge number i get the IVR but when i type the passcode for the bridge its not going through.

Any help would be highly appreciated.

2

you might want to set DTMF=inband instead of DTMF=rfc2833 i find this works better for external IVR’s
-Christopher

3

You should ask you service provider (who is terminating your PSTN call) what is the right format for DTMF.
Generally there are only 2 options - INFO and rfc2833.
‘Inband’ is transferred as audio, so it has no relation to compatibility between your * and your provider.

4

yes but sometimes Inband is the only way to get DTMF through to an outside Auto attendant… ive found some providers that wont send DTMF after the call is connected so with inband the DTMF is sent as audio and is generated at the phone itself and not in the switching network…
-Christopher

5

Hi,

Still have the same problem after the DTMF to inband, will also talk tot eh provide about the DTMF that i have to send from my side.

I was able to login to the bridge number only once, my passcode starts with 4 and when i dial 4 i get a message " wait your call is being transfered" and i get a error message.

Logs:
– SIP/provide.com-082fc680 answered SIP/cc100-0830c678
– Started music on hold, class ‘default’, on channel ‘SIP/provider.com-082fc680’
– Playing ‘pbx-transfer’ (language ‘en’)
– Stopped music on hold on SIP/provider.com-082fc680
– Unable to find extension ‘4’ in context ‘default’
– Playing ‘pbx-invalid’ (language ‘en’)
– Started music on hold, class ‘default’, on channel ‘SIP/provider.com-082fc680’
– Playing ‘pbx-transfer’ (language ‘en’)
– Stopped music on hold on SIP/provider.com-082fc680
– Unable to find extension ‘’ in context ‘default’
– Playing ‘pbx-invalid’ (language ‘en’)
== Spawn extension (default, tel number, 2) exited non-zero on ‘SIP/cc100-0830c678’
– Executing DeadAGI(“SIP/cc100-0830c678”, “call_log.agi|h”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
– AGI Script call_log.agi completed, returning 0
– Executing DeadAGI(“SIP/cc100-0830c678”, “VD_hangup.agi|h”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
– AGI Script VD_hangup.agi completed, returning 0

dial Plan:

exten => 1XXXNXXXXXX,1,Monitor(wav,${TIMESTAMP}${CALLERIDNUM}_${EXTEN},mb);
exten => _1XXXXXXXXXX,2,Dial(SIP/${EXTEN}@provider.com,60,tTO);

6

Asterisk 1.2.X’s DTMF is baly broken. Try applying this patch: h6315.com/dtmf.patch

7

Hi,

I am able to dial the passcode and get into the bridge, i have changed the DTMF to INFO as suggested by Andew.

Thanks to everybody for helping me out.