Queue crashes Asterisk

Hi all,

I’ve configured one queue in Asterisk (1.2.9.1 on Debian GNU/Linux w/ chan_misdn & chan_sccp).
When someone rings in, all phones in this queue rings, the caller is listening music. When one try to answer the call and picks up the phone, asterisk crashes.

Following output from asterisk (-vvvgc):
*CLI>
– SEP0017596ea926: New call on line 14
– SEP0017596ea926: Cisco Digit: 00000000 (0) on line 14
– Executing Answer(“SCCP/14-00000001”, “”) in new stack
– SCCP: Outgoing call has been answered SCCP/14-00000001 on 14@SEP0017596ea926-1
– Executing Wait(“SCCP/14-00000001”, “1”) in new stack
– Executing Queue(“SCCP/14-00000001”, “etrix”) in new stack
– Started music on hold, class ‘default’, on channel ‘SCCP/14-00000001’
– SEP001818856cfc: Asterisk request to call SCCP/10-00000002
– Called SCCP/10
– SEP001759a3c8cb: Asterisk request to call SCCP/12-00000003
– Called SCCP/12
– SEP0017596ea926: Asterisk request to call SCCP/14-00000004
– Called SCCP/14
– ATA1759e9628c01: Asterisk request to call SCCP/19-00000005
– Called SCCP/19
– SCCP/19-00000005 is ringing
– SCCP/14-00000004 is ringing
– SCCP/12-00000003 is ringing
– SCCP/10-00000002 is ringing
– SEP001818856cfc: Taken Offhook
– SEP001818856cfc: Answer the channel 10-2
– SCCP/10-00000002 answered SCCP/14-00000001
Ouch … error while writing audio data: : Broken pipe
Warning, flexibel rate not heavily tested!
Speicherzugriffsfehler (core dumped)

Here the relevant parts of my config files:
extensions.conf:
exten => 0,1,Answer
exten => 0,2,Wait,1
exten => 0,3,Queue(etrix)
exten => 0,4,Hangup

exten => 10,1,Dial(SCCP/10)
exten => 11,1,Dial(SCCP/11)
exten => 12,1,Dial(SCCP/12)
exten => 13,1,Dial(SCCP/13)
exten => 14,1,Dial(SCCP/14)
exten => 15,1,Dial(SCCP/15)
exten => 16,1,Dial(SCCP/16)
exten => 17,1,Dial(SCCP/17)
exten => 19,1,Dial(SCCP/19)
exten => 40,1,Dial(SCCP/40)

queues.conf:
[general]
;persistentmembers = yes

[etrix]
music=default
strategy=ringall
timeout=15
retry=5
wrapuptime=0
maxlen = 0
announce-frequency = 0
announce-holdtime = no

member => SCCP/10
member => SCCP/11
member => SCCP/12
member => SCCP/13
member => SCCP/14
member => SCCP/15
member => SCCP/16
member => SCCP/17
member => SCCP/19

Anyone knows what is going wrong here?
Regards,

-Andreas.

Search for ‘error while writing audio data’ in the forum.
You would find some threads where the user had faced the same problem.
Try the instructions in those threads.

the error is probably because asterisk doesn’t have an inbuilt audio player
and uses external player like mpg123 or others.

Hi vinod.vijayan,

I think, ‘error while writing audio data’ is not the problem. This is only a follow-up from the asterisk-crash.
I hear the MOH until one is taking the call and picks up the phone. Then asterisk wents down.

-Andreas.

what codecs are the phones using?

also, can you test with two SIP channels instead of SCCP? i ask only because i know we’ve had some issues getting SCCP working for our 7920’s and would be interested if the problem is in the queue or chan_sccp.

I’m using Cisco 7970 Phones and ATA186s. Do you think it’s better to load SIP onto the phones/ATAs? Is it worth the effort to develop&test the new config-files?

Not sure which codec is being used. sccp show device brings the following:
asterisk*CLI> sccp show device SEP0017596ea926
Current settings for selected Device

MAC-Address : SEP0017596ea926
Protocol Version : phone=9, channel=3
Keepalive : 60
Registration state : Ok(3)
State : OnHook(0)
MWI handset light : OFF
Description : Andi
Config Phone Type : 7970
Skinny Phone Type : Cisco7970(30006)
Softkey support : Yes
Autologin : 14
Image Version :
Timezone Offset : 0
Capabilities : 0x10c (ulaw|alaw|g729)
Codecs preference : (alaw|ulaw)
Can DND : Reject
Can Transfer : Yes
Can Park : No
Private softkey : Enabled
Can CFWDALL : Yes
Can CFWBUSY : Yes
Dtmf mode : In-Band
Trust phone ip : No
Early RTP : No
asterisk*CLI>
Lines
id : name label

1: 14 Andi (14)

i didn’t mean to replace your existing phones, just try a couple of softphones - Xlite is free and works pretty good, or there is idefisk for IAX.

it’s worth it to figure out if the problem is chan_sccp (since it’s technically a third party module) or app_queue.

Sorry, I misunderstood this :wink:

OK, I’ve idefisk here and give it a try.

Now I tested a SIP queue (with kphone) with the same result:
– Registered SIP ‘sipphone1’ at 192.168.1.211 port 5060 expires 900
– SEP0017596ea926: New call on line 14
– SEP0017596ea926: Cisco Digit: 00000006 (6) on line 14
– SEP0017596ea926: Cisco Digit: 00000001 (1) on line 14
– Executing Answer(“SCCP/14-00000001”, “”) in new stack
– SCCP: Outgoing call has been answered SCCP/14-00000001 on 14@SEP0017596ea926-1
– Executing Queue(“SCCP/14-00000001”, “etrix”) in new stack
– Started music on hold, class ‘default’, on channel ‘SCCP/14-00000001’
– Called SIP/sipphone1
– SIP/sipphone1-0fe4 is ringing
– SIP/sipphone1-0fe4 answered SCCP/14-00000001
Ouch … error while writing audio data: : Broken pipe
Warning, flexibel rate not heavily tested!
Speicherzugriffsfehler (core dumped)

Here the relevant config files:

queues.conf:
[etrix]
musiconhold=default
strategy=ringall
timeout=15
retry=5
wrapuptime=0
maxlen = 0
announce-frequency = 0
announce-holdtime = no

member => SIP/sipphone1
member => SIP/sipphone2

extensions.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[global]

[sccp]

exten => 61,1,Answer()
exten => 61,n,queue(etrix)

sip.conf:
[general]
context=sccp
language=de

[sipphone1]
type=friend
secret=xxx
qualify=yes
nat=no
host=dynamic
caninvite=no
context=sccp

[sipphone2]
type=friend
secret=xxx
qualify=yes
nat=no
host=dynamic
caninvite=no
context=sccp

this may sound dumb, but it’s worked for me in the past - do a clean recompile of asterisk and try again…

if that doesn’t work, you may want to look at trying a slightly older version of asterisk, such as 1.2.7.1

I’ve compiled now asterisk 1.2.10 with all channels (chan_capi, chan_misdn, chan_sccp) I have and restarted the machine, just to be sure.

But the problem still exists. I’m running out of ideas…

-Andreas.

I have the same problem with * 1.2.13 thus there is no problem with * but probably with audio(?) Idefisk(?) or what?