Summary:ASTERISK-07392: Unmute through *1 no longer works
Reporter:Michael Shuler (mikes2277)Labels:
Date Opened:2006-07-24 22:32:14Date Closed:2006-09-20 10:56:10
Versions:Frequency of
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.


 == 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')

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.

;  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, 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.