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Summary:ASTERISK-20592: no sound between chan_motif and psi
Reporter:Dmitry Melekhov (slesru)Labels:
Date Opened:2012-10-22 05:02:22Date Closed:2012-11-08 15:29:21.000-0600
Priority:MinorRegression?
Status:Closed/CompleteComponents:Channels/chan_motif
Versions:11.0.0-beta2 Frequency of
Occurrence
Related
Issues:
Environment:Centos 5/x86Attachments:( 0) motif.conf
( 1) motif.dump
( 2) myDebugLog
( 3) myDebugLog.11.0
( 4) myDebugLog-fromdahdi
( 5) xmpp.conf
Description:Hello!

Really, version I'm testing is rc2.
I call to meetme:
[jabber-in]                                                                                                                                  
exten => s,1,NoOp( Call from Gtalk )                                                                                                          
;exten => s,n,Answer                                                                                                                          
exten => s,n,Set(CALLERID(name)=”From Google Talk”)                                                                                          
;exten => s,n,Playback(/var/lib/asterisk/sounds/gena)                                                                                        
;exten => s,n,Dial(DAHDI/g1/6401,,g)                                                                                                          
exten => s,n,Meetme(6000,TL(10800000:60000))                                                                                                  
exten => s,n,Hangup      


And hear nothing.

only speex wide band is allowed:
disallow=all                                                                allow=speex16                      

I can call another psi just fine.

And I'll upload traffic dump.
22.19 is asterisk, 22.229 is my desktop

Comments:By: Dmitry Melekhov (slesru) 2012-10-22 05:02:43.948-0500

traffic dump
By: Rusty Newton (rnewton) 2012-10-29 15:52:44.755-0500

Dmitry,

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

We need additional information to look into this issue.

Please provide xmpp.conf, motif.conf, the version and OS of your XMPP client (PSI?)

Please provide a full log captured during the entire call that demonstrates the issue. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
Be sure that VERBOSE and DEBUG are set to at least level 5. Before making the call be sure that "xmpp set debug on" is enabled and then perform "logger reload"

As you have done with the dump, please attach the files to the issue.

By: Dmitry Melekhov (slesru) 2012-10-31 23:39:04.765-0500

log contains sip calls, not only xmpp call, sorry :-(
By: Dmitry Melekhov (slesru) 2012-10-31 23:41:28.415-0500

Hello!

I upgraded my desktop to Ubuntu 12.10/x86-64 ( from 12.04 ), psi+ was upgraded to, no asterisk do not answer to call, the same is for call from asterisk to psi ( I'll debug this call later, with no users).

xmpp client now is Psi+ v0.16-dev-20121030.

Thank you!


By: Dmitry Melekhov (slesru) 2012-11-01 03:56:48.384-0500

Call from dahdi to asterisk.
From psi+ point of view connection is established, but I still hear call tone ( don't know english term) on dahdi side.
By: Dmitry Melekhov (slesru) 2012-11-01 22:45:22.505-0500

btw, situation is stranger than I thought.

just tried to reproduce situation.

I restarted asterisk and got call connected, but no sound:

this is what I see on console:
   -- Executing [s@jabber-in:1] NoOp("Motif/dm-2f1a", " Call from Gtalk ") in new stack
   -- Executing [s@jabber-in:2] Answer("Motif/dm-2f1a", "") in new stack
   -- Executing [s@jabber-in:3] Set("Motif/dm-2f1a", "CALLERID(name)=”From Google Talk”") in new stack
   -- Executing [s@jabber-in:4] MeetMe("Motif/dm-2f1a", "6000,TL(10800000:60000)") in new stack
 == Parsing '/etc/asterisk/meetme.conf': Found
   -- Created MeetMe conference 1023 for conference '6000'
   -- Setting conference duration limit to: 10800000ms.
   -- Setting warning time to 60000ms from the conference duration limit.
   -- <Motif/dm-2f1a> Playing 'conf-onlyperson.gsm' (language 'en')

Tried to connect 2 or 3 times, result is the same, so no regression in newer psi+.

Then I set verbose, debug to 15 and xmpp set debug on.

From psi point of view call is not established,  but from asterisk point of view it is established:
   -- Executing [s@jabber-in:3] Set("Motif/dm-2dd2", "CALLERID(name)=”From Google Talk”") in new stack
   -- Executing [s@jabber-in:4] MeetMe("Motif/dm-2dd2", "6000,TL(10800000:60000)") in new stack
 == Parsing '/etc/asterisk/meetme.conf': Found
   -- Created MeetMe conference 1023 for conference '6000'
   -- Setting conference duration limit to: 10800000ms.
   -- Setting warning time to 60000ms from the conference duration limit.
   -- <Motif/dm-2dd2> Playing 'conf-onlyperson.gsm' (language 'en')



asterisk*CLI> core show channels
Channel              Location             State   Application(Data)            
DAHDI/pseudo-1743501 s@default:1          Rsrvd   (None)                        
Motif/dm-2dd2        s@jabber-in:4        Up      MeetMe(6000,TL(10800000:60000)
2 active channels
1 active call
1 call processed

tried several times and got very slow system , LA about 10, had to restart asterisk.

Looks like there is problem in signalling too :-(



By: Joshua C. Colp (jcolp) 2012-11-02 06:11:10.522-0500

Are you still testing with RC2? There were 2-3 changes related to Motif done for the release.
By: Dmitry Melekhov (slesru) 2012-11-02 06:13:56.641-0500

Hello!

Yes, yes still -rc2, I'll test release next week and report about results.

Thank you!

By: Dmitry Melekhov (slesru) 2012-11-05 21:45:44.255-0600

the same issue with 11.0, here is log which demonstrates this, it contains 3 calls.
Firts call passed, but no sound, second was rejected by asterisk, third stuck- from asterisk point of view it is connected, from psi+ it is not.

By: Joshua C. Colp (jcolp) 2012-11-06 05:55:23.613-0600

What XMPP server are you using? It looks as though it is disconnecting us which may be causing traffic to get lost.
By: Dmitry Melekhov (slesru) 2012-11-06 06:03:43.434-0600

I use ejabberd, namely 2.1.10-2ubuntu1 .
As I wrote before psi - psi calls work OK with the same server...


By: Joshua C. Colp (jcolp) 2012-11-06 06:06:30.881-0600

Can you look at the ejabberd log to see if it says why it is disconnecting Asterisk?
By: Dmitry Melekhov (slesru) 2012-11-06 06:15:35.752-0600

There is no info about disconnection.
Just tried, last info in ejabberd log is :


=INFO REPORT==== 2012-11-06 07:40:12 ===
I(<0.30820.1>:ejabberd_c2s:938) : ({socket_state,tls,{tlssock,#Port<0.21040140>,#Port<0.21040142>},<0.30819.1>}) Opened session for asterisk@jabber.belkam.com/asterisk-xmpp

=INFO REPORT==== 2012-11-06 07:40:32 ===


right now:

date
Вт. нояб.  6 16:15:25 MSK 2012


By: Joshua C. Colp (jcolp) 2012-11-06 06:56:46.284-0600

I've isolated the problem to Psi *requiring* that the payload name for speex be capitalized, despite the RFC for it saying otherwise (http://tools.ietf.org/html/rfc5574). If the payload name is not capitalized it will happily send Asterisk media and receive media, but not send it to the sound card. Change it to be capitalized and boom - it's happy and you can hear audio. I'll see about what can be done for that...
By: Dmitry Melekhov (slesru) 2012-11-06 09:56:12.071-0600

Hello!

Thank you!

I posted information about psi incompatibility to their mail list.
http://lists.affinix.com/pipermail/psi-devel-affinix.com/2012-November/009358.html

Hope psi developers will tell me what can I do.

Thank you!
By: Dmitry Melekhov (slesru) 2012-11-06 23:37:27.566-0600

As I understand psi send and receive payload with name SPEEX instead of speex.
So setting it to speex solves psi receiving problem.
But will asterisk receive payload with name SPEEX? I don't hear sound on asterisk side too...
May be it is possible to at least receive such not properly named payload?

By: Joshua C. Colp (jcolp) 2012-11-07 05:44:22.526-0600

Asterisk performs a case insensitive comparison, and seems to be at no fault with receiving audio. I've confirmed this. That doesn't mean Psi is actually sending real audio, though.
By: Dmitry Melekhov (slesru) 2012-11-07 05:49:45.435-0600

Thank you!

So this is psi bug and should be fixed in psi.

I already got reply from psi developer and he is going to fix this.

Thank you!

By: Matt Jordan (mjordan) 2012-11-08 15:29:21.231-0600

Closing out as "not a bug", since the issue apparently lies with psi.