[Home]

Summary:ASTERISK-25066: DTMFs are not sent to the bridge channel if they are used by any built-in or dynamic feature
Reporter:Etienne Allovon (etienne_pf)Labels:
Date Opened:2015-05-06 08:26:37Date Closed:2017-12-18 09:10:07.000-0600
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General Features
Versions:11.17.1 Frequency of
Occurrence
Constant
Related
Issues:
Environment:Attachments:( 0) ASTERISK-25066-asterisk-full_extract.log
( 1) ASTERISK-25066-sip.conf
( 2) ASTERISK-25066-trace.pcap
Description:Gvien the following features.conf :
{code}
[featuremap]
blindxfer = *1
disconnect = *0
automon = *3
atxfer = *2
{code}


If I call an outbound IVR (with the Dial option T)
If I press * the * is not sent to the bridge channel unless you press another key after that.
But the next digit is never sent.

For example, If I press 1, *, 7 the IVR will get 1 and * :
{code}
PhoneA     asterisk     Phone B
1 ---->---- 1
            1 --->--- 1
* ---->---- *
7 ---->---- 7
            * --->--- *
{code}
Comments:By: Etienne Allovon (etienne_pf) 2015-05-06 08:58:44.339-0500

Log full with
core set debug 5

See :
line 26, asterisk interprets the * as the beginning of a feature
line 27 : it sets a timeout of 1500ms
line 27 onward : no DTMF is sent (no line like _Trying to put 'INFO sip:im'_)
line 39 : the features timeout ends long after the 1500ms, i.e. *when the next DTMF is received*
line 42 : the DTMF is sent but it is the *, not the 7 and the 7 is never sent

By: Rusty Newton (rnewton) 2015-05-11 17:57:19.347-0500

Can you attach a SIP packet trace captured during the issue, along with the configuration for the SIP peer?

By: Etienne Allovon (etienne_pf) 2015-05-12 02:41:07.209-0500

Here's the SIP trace.

*Test :*
PhoneA (cobs82, 10.32.1.13) calls PhoneB (kyb55w, 10.32.1.17) via asterisk (10.32.1.2).
PhoneB answers
PhoneA sends DTMFs 1 * 7 with 5 seconds pause between each DTMF

*Result :*
asterisk sends only 1 and * to PhoneB

*Configuration :*
{code}
[featuremap]
blindxfer = *1
disconnect = *0
automon = *3
atxfer = *2
{code}

sip.conf (see attached file).

dialplan :
{code}
exten = 1010,1,NoOp()
same = n,Dial(SIP/kyb55w,,T)
{code}

By: Joshua C. Colp (jcolp) 2017-12-18 09:10:07.719-0600

The way that DTMF based features in bridging work was changed massively. After looking at this issue I do not believe it is applicable to current support versions.

By: Etienne Allovon (etienne_pf) 2018-01-04 09:28:34.754-0600

I confirm that it doesn't show up anymore in asterisk 13

By: Asterisk Team (asteriskteam) 2018-01-04 09:28:35.158-0600

This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable.