Summary:	ASTERISK-07392: Unmute through *1 no longer works
Reporter:	Michael Shuler (mikes2277)	Labels:
Date Opened:	2006-07-24 22:32:14	Date Closed:	2006-09-20 10:56:10
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Applications/app_meetme
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	When pressing *1 when the admin it tells you that you are unmuted but no one can hear you.  I have tested this with ztdummy.  I do not know if it has the same issue if you have a real zaptel card. ****** ADDITIONAL INFORMATION ****** pbx1*CLI>  == Spawn extension (barge, 15086, 0) exited non-zero on 'SIP/5060-08fd4098' in macro 'dial'  == Spawn extension (barge, 15086, 0) exited non-zero on 'SIP/5060-08fd4098' in macro 'exten-vm'  == Spawn extension (barge, 15086, 0) exited non-zero on 'SIP/5060-08fd4098'    -- Executing MeetMe("SIP/5060-08fd4098", "200000005086|qd|271721") in new stack    -- Created MeetMe conference 1023 for conference '200000005086'    -- Executing MeetMe("SIP/5086-0900f6b8", "200000005086|qd|271721") in new stack       > Channel SIP/5076-09069550 was answered.    -- Executing MeetMe("SIP/5076-09069550", "200000005086|aqs|271721") in new stack    -- Playing 'conf-adminmenu' (language 'en')    -- Playing 'conf-unmuted' (language 'en') pbx1*CLI>
Comments:	By: Joshua C. Colp (jcolp) 2006-07-26 11:06:59 I was unable to recreate this issue using the latest 1.2 checkout. Could it be a sign of NAT troubles instead? By: Michael Shuler (mikes2277) 2006-07-26 18:39:41 Both phones are behind a NAT firewall but they are both able to make outbound and inbound calls without issue.  I will try without the NAT tomorrow and let you know what I find.  Thanks! By: Michael Shuler (mikes2277) 2006-08-01 10:39:35 Sorry it took me so long, I got pulled off onto other projects.   I tried it (all non NAT) using 2 PBX phones and then a 3rd to barge with that were all on the same subnet and all registed directly with the same Asterisk server.  I tried a mixture of Grandstream GXP-2000's and Linksys SPA-942's but the result was the same.  When pressing *1 Asterisk would tell me that I am unmuted but no one could hear me.  When pressing *1 again, it would say I was unmuted (and again, and again).  It never would say that I was muted after pressing *1. By: Michael Shuler (mikes2277) 2006-08-01 11:11:11 I also updated to today's SVN (38611) which did not help. By: Joshua C. Colp (jcolp) 2006-08-01 11:15:40 This is just something I noticed when figuring out your meetme arguments but the one that sends the person in as an admin doesn't have the d argument for dynamic, if you add that in does it work? It's small but might make a difference. If the problem still occurs I'll lab it up using the exact settings you have. Also - could you post the Meetme lines here so I can confirm? By: Joshua C. Colp (jcolp) 2006-08-08 12:06:23 It's been a week and still no reply - if this is still an issue reopen this bug with the configuration files or catch me on IRC in #asterisk-bugs as file for live debugging. Thanks! By: Michael Shuler (mikes2277) 2006-08-08 13:26:06 Sorry I missed the last email notification on the ticket update.  It still doesn't work with the "d" added. [barge] ; ;  support the ability of barging into calls ; exten => _1X.,1,MeetMe(20000000${EXTEN:1},qd,271721) exten => _2X.,1,MeetMe(20000000${EXTEN:1},aqds,271721) By: Joshua C. Colp (jcolp) 2006-08-08 14:07:56 I just tested that on latest 1.2 and trunk, both are fine. I think next we should move onto the actual SIP connection to rule out issues with that. Can you do an rtp debug so I can confirm no audio issues exist? By: Jason Parker (jparker) 2006-08-14 11:28:18 Can you provide the rtp debug as requested? By: Serge Vecher (serge-v) 2006-08-25 11:22:57 no response from mikes2277, again ... By: Michael Shuler (mikes2277) 2006-08-27 21:32:51 My spam filters must be getting these because I just got a notification stating that I had not responded.  Anyway, how do you want me to make an RTP debug for you? i.e. tcpdump or something else within Asterisk. By: Joshua C. Colp (jcolp) 2006-08-30 10:53:02 Just type rtp debug on the Asterisk CLI, it'll print all RTP activity. By: Serge Vecher (serge-v) 2006-09-20 10:56:09 no response for two weeks again. mikes2277, if you are still having a problem in 1.2.12.1, please join the #asterisk channel on IRC and attempt to debug with somebody "live." If the problem is indeed a bug in Asterisk, please reopen with pertinent information attached.