[Solved] IAX member cannot disconnect when using queues

Asterisk Asterisk Support
1

Hi I configured Skype Connect to receive calls to my asterisk box. When I set dial plan to dial directly IAX member it works fine.
But when I put member on queue and he answers call I get connection without sound and lots of warnings. here is log from call:

== Using SIP RTP CoS mark 5 -- Executing [<skypein_phone_no>@from-skype:1] NoOp("SIP/990510xxxxxxxx-00000000", " , 353871719696") in new stack -- Executing [<skypein_phone_no>@from-skype:2] Goto("SIP/990510xxxxxxxx-00000000", "queues-from-customer,s,1") in new stack -- Goto (queues-from-customer,s,1) -- Executing [s@queues-from-customer:1] Answer("SIP/990510xxxxxxxx-00000000", "") in new stack -- Executing [s@queues-from-customer:2] Queue("SIP/990510xxxxxxxx-00000000", "queue.pl-en") in new stack -- Started music on hold, class 'default', on SIP/990510xxxxxxxx-00000000 -- Call accepted by 178.167.131.154 (format gsm) -- Format for call is gsm -- IAX2/t0002-4356 is ringing > doing dnsmgr_lookup for 'sip.skype.com' > ast_get_srv: SRV lookup for '_sip._udp.sip.skype.com' mapped to host 3.sip.skype.com, port 5060 -- IAX2/t0002-4356 answered SIP/990510xxxxxxxx-00000000 [Jan 18 20:49:43] WARNING[22905]: channel.c:1476 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/t0002-4356 [Jan 18 20:49:45] WARNING[22909]: channel.c:1476 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/t0002-4356 [Jan 18 20:49:46] WARNING[22903]: channel.c:1476 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/t0002-4356 [Jan 18 20:49:47] WARNING[22907]: channel.c:1476 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/t0002-4356 [Jan 18 20:49:48] WARNING[22910]: channel.c:1476 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/t0002-4356 [Jan 18 20:49:50] WARNING[22910]: channel.c:1476 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/t0002-4356 > doing dnsmgr_lookup for 'sip.skype.com' > ast_get_srv: SRV lookup for '_sip._udp.sip.skype.com' mapped to host 3.sip.skype.com, port 5060
At this point I’m breaking connection with my member, I’m still conencted on skype end, and have to close connection manually.

[Jan 18 20:50:21] WARNING[22903]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 11, ts=30005, seqno=6)
[Jan 18 20:50:31] WARNING[22909]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 11, ts=40004, seqno=7)
[Jan 18 20:50:33] WARNING[22904]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 2, ts=42005, seqno=8)
       > doing dnsmgr_lookup for 'sip.skype.com'
       > ast_get_srv: SRV lookup for '_sip._udp.sip.skype.com' mapped to host 3.sip.skype.com, port 5060
[Jan 18 20:50:41] WARNING[22903]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 11, ts=50004, seqno=9)
[Jan 18 20:50:49] WARNING[22915]: chan_sip.c:3866 __sip_autodestruct: Autodestruct on dialog 'CXC-311-65a66a50-44da78c1-13c4-4f17306e-f16117aa-3fc014f8' with owner in place (Method: BYE)
[Jan 18 20:50:51] WARNING[22909]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 11, ts=60004, seqno=10)
[Jan 18 20:50:54] WARNING[22904]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 2, ts=63004, seqno=11)
[Jan 18 20:51:01] WARNING[22908]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 11, ts=70003, seqno=12)
       > doing dnsmgr_lookup for 'sip.skype.com'
       > ast_get_srv: SRV lookup for '_sip._udp.sip.skype.com' mapped to host 3.sip.skype.com, port 5060
[Jan 18 20:51:11] WARNING[22904]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 11, ts=80003, seqno=13)
[Jan 18 20:51:15] WARNING[22909]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 2, ts=84004, seqno=14)

at this point I close member’s soft phone disconnecting it from asterisk. But it still is visible as connected peer

atademo*CLI> iax2 show peers
Name/Username    Host                 Mask             Port          Status    
t0001/t0001      (null)          (D)  255.255.255.255  0             Unmonitored
t0003/t0003      (null)          (D)  255.255.255.255  0             Unmonitored
t0002/t0002      178.167.131.154 (D)  255.255.255.255  4569          Unmonitored
t0004/t0004      (null)          (D)  255.255.255.255  0             Unmonitored
4 iax2 peers [0 online, 0 offline, 4 unmonitored]
[Jan 18 20:51:21] WARNING[22903]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 11, ts=90003, seqno=15)
atademo*CLI> iax2 reload
  == Parsing '/etc/asterisk/iax.conf':   == Found
[Jan 18 20:51:30] NOTICE[22929]: chan_iax2.c:13018 set_config: Ignoring bindport on reload
    -- Seeding 't0002' at 178.167.131.154:4569 for 120
[Jan 18 20:51:31] WARNING[22909]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 11, ts=100003, seqno=16)
[Jan 18 20:51:36] WARNING[22904]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 2, ts=105003, seqno=17)
       > doing dnsmgr_lookup for 'sip.skype.com'
       > ast_get_srv: SRV lookup for '_sip._udp.sip.skype.com' mapped to host 3.sip.skype.com, port 5060
[Jan 18 20:51:41] WARNING[22908]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 178.167.131.154 on IAX2/t0002-4356 (type = 6, subclass = 11, ts=110003, seqno=18)
atademo*CLI> 
Disconnected from Asterisk server
Executing last minute cleanups
engrost@atademo:~$ sudo /etc/init.d/asterisk restart

The only resolution to stop those ‘__attempt_transmit: Max retries exceeded’ is to restart server. I’m fairly newbe to sterisk so I’m probably doing sth wrong. Please help.
I’m using asterisk :
Asterisk 1.8.8.1-1digium1~natty, Copyright © 1999 - 2011 Digium, Inc. and others.
below are my config files:
iax.conf

[general]
bindport=4569
iaxcompat=yes
bandwidth=medium
disallow=lpc10                  ; Icky sound quality...  Mr. Roboto.
jitterbuffer=no
forcejitterbuffer=no
minregexpire = 120
maxregexpire = 120
autokill=yes
calltokenoptional = 0.0.0.0/0.0.0.0

[t0001]
type=friend
username=t0001
secret=<my_secret>
host=dynamic
context=queueagent
nat=yes

[t0002]
type=friend
username=t0002
secret=<my_secret>
host=dynamic
context=queueagent
nat=yes

[t0003]
type=friend
username=t0003
secret=<my_secret>
host=dynamic
context=queueagent
nat=yes

[t0004]
type=friend
username=t0004
secret=<my_secret>
host=dynamic
context=queueagent
nat=yes

queues.conf

[general]

persistentmembers = yes
monitor-type = MixMonitor
updatecdr = yes
[testqueue]
musicclass=default
announce=Test Queue
strategy=leastrecent
timeout=15
retry=5
timeoutpriority=conf
ringinuse=no
;joinempty=paused,inuse,invalid

member=>IAX2/t0001
member=>IAX2/t0002
member=>IAX2/t0003
member=>IAX2/t0004

extensions.conf

[general]
static=yes
writeprotect=no
autofallthrough=no

[globals]
TRUNK=Skype

[from-skype]
;for calling from skypein number
exten => xxxxxxxxxxxx,1,Noop(${CALLERID(name)} , ${CALLERID(num)})
exten => xxxxxxxxxxxx,n,Goto(queues-from-customer,s,1)

;for calls from skype
exten => 9905100xxxxxxx,1,Noop(${CALLERID(name)} , ${CALLERID(num)})
exten => 9905100xxxxxxx,n,Goto(queues-from-customer,s,1)


[queues-from-customer]
exten => s,1,Answer
exten => s,2,Queue(testqueue)

Note: I have hidden some private info with ‘x’

Edit:
I’m sorry I posted is as deadlock, but after reading this: voip-info.org/wiki/index.php?page_id=741
I came to conclusion that my situation is not deadlock after all.

2

wiki.asterisk.org/wiki/display/ … rADeadlock

You may need to compile from source or request a debug version from the packager.

3

I do not think that is deadlock anymore sorry for confusion.
As I said I’m fairly new to asterisk and now I’m just trying to prepare this demo system that handles queues, with IAX members. All calls comming from skype connect and it just does not work as I thought it would be.

4

After a lot of digging on bugs and scraps of information I found solution. That I dunno but it works
after changing codec to ulaw in iax.conf
disallow=all
allow=ulaw

everything works as it should.