Summary:	ASTERISK-10251: Audio problems (one direction) with 2 SIP peers when using queues and DTMF tones
Reporter:	daphi (daphi)	Labels:
Date Opened:	2007-09-07 08:08:27	Date Closed:	2011-06-07 14:03:23
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) sip_321_debug.txt
Description:	Given the standard sip.conf, which is was only modified as specified below. The sipgate peer is used for incoming calls only and is routed into a queue. The main problems is, when the caller sends a dtmf tone the callee cannot hear the caller anymore, although the caller can hear the callee. If the sipgate extension is routed directly to the sip phone (321) without going into the queue first there is no problem with dtmf tones. After the dtmf tone is transmitted both parties can hear each other again. So the problem must be related to queue. We have tested the same setup on Asterisk 1.2.10 without any issues. ****** ADDITIONAL INFORMATION ****** -------- sip.conf --------- [general] disallow=all allow=alaw allow=ulaw dtmfmode=inband and the following sip entries: register => 12345:xxxxx@sipgate.at/12345 [321] type=friend user=321 secret=xxx callerid=xxx <321> host=dynamic mailbox=321 -------- sip.conf --------- ------- queues.conf ------- [ws1] musiconhold = ws1 strategy = roundrobin timeout = 15 retry = 1 reportholdtime = no wrapuptime = 0 maxlen = 10 weight = 1 ringinuse = no announce-frequency = 0 announce-holdtime = no joinempty=strict timeoutrestart = yes memberdelay = 0 member => Agent/611,1 member => Agent/612,2 member => Agent/613,3 ------- queues.conf -------
Comments:	By: Mark Michelson (mmichelson) 2007-09-07 09:15:18 I could not reproduce this issue locally using the 1.4's latest SVN checkout. By: daphi (daphi) 2007-09-10 11:45:50 Well, it seems it was our mistake. There was a misconfiguration between the DTMF modes set by asterisk an the sip phone we are using. Asterisk has to be set to 'rfc' dtmf mode and the pones (Linksys) have to be set to dtmf mode 'auto'. Case can be closed. By: Michiel van Baak (mvanbaak) 2007-09-10 11:57:16 thanks for reporting back.