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Summary:ASTERISK-13260: Using dtmfmode=info AND canreinvite=yes (both in sip.conf) AND dynamic features (features.conf/Dial() with w or W flags)
Reporter:Michel Belleau (malaiwah)Labels:
Date Opened:2008-12-22 15:22:34.000-0600Date Closed:2011-06-07 14:07:20
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
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Description:If Asterisk is configured for SIP INFO and re-invites, if one wants to enable dynamic features on a Dial application call, Asterisk will stay in the audio path for all the conversation.

This behavior should be correct for in-band DTMF and RFC2833 but for SIP INFO, it could still send a re-invite and get back the voice path IF a dynamic feature is engaged (in my case, it is #3 for quick recording of the conversation).

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

My goal is to make Asterisk an Application Server/Call Dispatcher which can handle more than 125 conccurent calls per host (I think this was the core limit I have red somewhere earlier, 250 call legs which means 125 natively bridged calls) by NOT HAVING Asterisk constantly in the media path.

SIP INFO is one of the things that should be enabled for this to happen.
Comments:By: Russell Bryant (russell) 2008-12-22 15:44:15.000-0600

I just want to note that your note about a built in limit for maximum number of calls is bogus.  There is no such limit.

Can you please provide your configuration files, a SIP trace of the call in question, as well as your console output when doing a test?  That will help us reproduce the problem.
By: Michel Belleau (malaiwah) 2008-12-22 16:00:30.000-0600

Hi Russell.

I was giving numbers by heart, but there are various evidences in the forums of conccurent call limits like this, but mainly this is what we find on voip-info.org (you might edit the page if this info is bogus, http://www.voip-info.org/wiki/index.php?page=Asterisk+dimensioning):

{{{
The only rule of thumb we appear to be able to provide is this: Asterisk 1.2 start to run into problems around 220 concurrent SIP calls. Asterisk 1.4 scales much better and can handle nearly double the call setups/second as well as total concurrent traffic. Moreover, Early testing of Asterisk 1.6 using hash tables shows a SIP performance increase compared to 1.4 of factor 3 to 4.
}}}

So 1.2=200 bridged calls, 1.4=400 bridged calls and 1.6=1200 bridged calls?

I might give a try on Asterisk 1.6 then, but anyway my case stand still Asterisk could re-invite audio paths when peer is using dtmfmode=info.

I will create a simple test case and post it back then.
By: Leif Madsen (lmadsen) 2009-01-06 08:26:38.000-0600

malaiwah, have you been able to obtain the information requested by Russell?
By: Leif Madsen (lmadsen) 2009-01-13 12:35:34.000-0600

This is a reminder for a request for information on an issue you have opened. It is policy to close issues after 7 days should a request for information go unanswered.

If you could get the information requested, then we can move the issue along for you.

Thanks!
By: Leif Madsen (lmadsen) 2009-02-10 15:04:47.000-0600

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