[Asterisk-Users] Gaps in sound

[Paul Rodan]

I upgraded to CVS Asterisk 1.0 stable last night on 2 different servers,
connected to each other via IAX2.

 

Voice T1/PRI -> Cisco 3640  -SIP-> Main Asterisk -IAX2-> Remote Asterisk
-SIP-> Phones

 

All using the g711ulaw codec. 

 

We're now experiencing gaps in sound, the other party will be talking to us
and suddenly they'll cut out and then come back in, less than a couple of
second's gap. Nothing on the CLI shows what could be causing this. Just
occurs out of nowhere. I don't think it's a connectivity issue, as
physically this is all at the same location. The Main Asterisk connects to
the Remote Asterisk via a private switch, and the Remote Asterisk connects
to the phones via another interface connected to the private LAN via a
switch.

 

 

I did this upgrade to get rid of a problem, there was the occasional call
where these "gaps" in sound would occur, but it was 1 out of 7 or something
like that. I upgraded last night, rebooted the server to clear up some
memory and force the reloading of the newest zaptel and libpri modules. Now
I'm running "Asterisk CVS-v1-0-11/16/04-04:38:19"

 

My only clue as to the problem may be this:

 

---

Nov 16 12:14:21 WARNING[1161]: channel.c:472 ast_channel_walk_locked:
Avoided initial deadlock for 'SIP/55.55.55.5-42d09bf0', 10 retries!

---

I see this a whole lot. I don't have a clue what it means. The
SIP/55.55.55.5 IP address has been changed to protect the innocent. The IP
that was after SIP/ is the IP of our Cisco 3640 router.

 

Any help would be greatly appreciated, as I'm getting really tired of VOIP.
Starting to loose my enthusiasm for it. Quality issues and bugs left and
right and screaming bosses and customers and high cell phone bills.

 

Regards,

Paul

 

 

P.S - 

 

Also, I see this now and then, but I think it's not related, as this has
nothing to do with the Remote Asterisk server I spoke of:

---

Nov 16 12:14:21 WARNING[1161]: ast_expr.y:474 ast_yyerror: ast_yyerror():
syntax error: parse error; Input:

"unknown" <55.55.55.5> = 1235551212

^^^^^^^^

---

 

The 55.55.55.5 Ip was changed from the real value, it was the IP address of
our Cisco 3640. Also, 555121212 was a real phone number, just changed it to
protect the innocent. The only references to this "1235551212" number is in
my dialplan:

 

exten => 1235551212,1,SetAccount(800-2323)

exten => 1235551212,2,SetGroup(customer)

exten => 1235551212,3,CheckGroup(5)

exten => 1235551212,4,GotoIf($[${CALLERID} = 9547726277]?40:5)          ;
Offending Caller

exten => 1235551212,5,GotoIf($[${CALLERIDNUM} = 9547726277]?40:6)

exten => 1235551212,6,Goto(customer-in,100,1)

exten => 1235551212,40,Answer                                           ;
Where to put offending caller

exten => 1235551212,41,Wait(2)

exten => 1235551212,42,Background(tt-weasels)

exten => 1235551212,43,Hangup

exten => 1235551212,104,VoiceMail(u100 at customer)                        ;
When they are out of lines

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