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Summary:ASTERISK-16108: DTMF (info) is no longer passed through after 180 seconds
Reporter:Jacco van Tuijl (jacco)Labels:
Date Opened:2010-05-17 08:36:35Date Closed:2011-06-07 14:04:41
Priority:MinorRegression?No
Status:Closed/CompleteComponents:General
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Description:DTMF (info) is no longer passed through after 180 seconds.
180 seconds after pickup DTMF is no longer passed trough to the other side.
The DTMF is recieved correctly by asterisk

also SendDTMF() doesn't send any DTMF

****** ADDITIONAL INFORMATION ******

I check this problem agains all 1.6.x versions : they all have the same problem
Comments:By: Jacco van Tuijl (jacco) 2010-05-17 08:49:15

everything works fine before the 180 seconds have passed.
It seems to be exacly 180 seconds.
By: Paul Belanger (pabelanger) 2010-05-17 10:52:57

We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue:

http://svn.digium.com/svn/asterisk/trun/doc/HOWTO_collect_debug_information.txt
By: Jacco van Tuijl (jacco) 2010-05-18 04:48:57

NAT issue

qualify does not sent nat-keepalive during the conversation for unregisterd phones that use the sip.conf [general] settings:
qualify=yes
qualifyfreq=60

maybe the description of this issue should be changed
By: Jacco van Tuijl (jacco) 2010-05-19 06:13:28

when I keep sending DTMF tones with an interval smaller then the 180seconds nat timeout, actualy doing a manual NAT keep-alive bij sending SIP-info packet, DTMF keep working perfectly even afther the 180seconds NAT timeout.

This is because qualify timers only start on registration and so will not send any nat keepalive to unregisterd phones.(i found this on the web somewhere; I did not verify)

I guess the timers should also be started when an unregisterd phone makes a call or makes a seperate timer for nat-keepalive that also works for unregisterd phones
By: Paul Belanger (pabelanger) 2010-05-19 10:48:13

This seems to be more of a NAT issue on your firewall then Asterisk. To my knowledge the only _real_ solutions in Asterisk is to lower your registration / qualify timers in an attempt to keepalive NAT.  Not the best solution, more of a workaround.

At this point, I'm not sure what else can be done in Asterisk, unless this feature is added.
By: Jacco van Tuijl (jacco) 2010-05-20 04:28:37

it is not a nat issue in my firewall; it's a nat issue in asterisk
Please read my comments and ask my to clearify if my details are not clear or if my english is bad.
This seems like an easy to solve problem to me and I thought my explenation is very clear.

"is to lower your registration / qualify timers "

The Qualify functionality is broken/does not work in some cases.

I'm allowing phone that did not register to make phonecalls.
I think the qualify timers should be started, for every phone that has set  qualify=yes,  as soon as it starts making a call if it hasn't been started already.
(now qualify timers are only started during registration ; because not every phone needs to be registerd this is not working properly in all cases )

Kind regards,
Jacco
By: Paul Belanger (pabelanger) 2010-05-20 08:56:23

Please follow below to create a debug log, then reproduce your issue.
--
We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue:

http://svn.digium.com/svn/asterisk/trun/doc/HOWTO_collect_debug_information.txt
By: Jacco van Tuijl (jacco) 2010-05-20 09:21:08

the link to HOWTO is 404
By: Paul Belanger (pabelanger) 2010-05-20 09:36:58

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
By: Paul Belanger (pabelanger) 2010-06-03 20:52:38

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.

Further information can be found at http://www.asterisk.org/developers/bug-guidelines