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Summary:ASTERISK-05006: overlapping ringing and voice
Reporter:sohosys (sohosys)Labels:
Date Opened:2005-09-06 10:05:48Date Closed:2005-09-13 23:08:23
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) debug.txt
Description:when sending a call originated on a SIP ATA to a SIP provider via Asterisk there is a delay in answer detection that results in an additional ring overlapping the voice in the first second of the call.

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

codec is g.711u from ata to asterisk and from asterisk to provider.
ATA is linksys (sipura), also tested with snom 190 phone.
same result with 2 providers (XO and Qwest), both using a Sonus softswitch (GSX9000?).
most noticable when reaching a voicmemail box where the greeting starts right after answer.

Comments:By: sohosys (sohosys) 2005-09-06 10:18:47

easier to reproduce when calling a cellular phone and reaching voicemail, sprint PCS and Verizon used to test.
By: Brian West (bkw918) 2005-09-06 10:58:31

This is fixed in the Sipura by setting Sticky 183 to YES.

/b
By: Brian West (bkw918) 2005-09-06 11:00:05

. Added <Sticky 183> boolean paramter under Ext 1/2/3/4. If set to
   "yes", SPA will ignore further 180 SIP responses after receiving the
   first 183 SIP response for an outbound INVITE. This avoids double
   ring-back tone in case the 180 response comes after 183 out-of-order

Chances are asterisk sends them too fast or in the wrong order.  Either case should be fixed.  But Sticky 183 fixes it on my Sipura!
By: Olle Johansson (oej) 2005-09-06 11:02:58

Can we get a SIP debug on this?
By: sohosys (sohosys) 2005-09-06 11:24:01

sip debug uploaded. will try the sipura "sticky 183" assuming the linksys gear has this option.
By: sohosys (sohosys) 2005-09-06 11:27:47

the linksys ATA does not appear to have a "sticky 183" option. Is there a trick I should know or is it obvious on the sipura config interface?
By: Kevin P. Fleming (kpfleming) 2005-09-13 23:08:08

This is a bug in your SIP ATA. Once it has received '183 Session Progress' and is receiving early audio, it must not start its own audio in response to any '180 Ringing' it may receive.