[asterisk-users] Asterisk hangup all incoming calls after 10 seconds

Tue Mar 16 14:55:18 CDT 2010

Hello Gentleman,

I'm new to asterisk, this is my first instalation, first post... so I'd like
to apologize if this question has already been made. I googled but I
couldn't find nothing similar.

Here's the thing.

I'm migrating from ATA to Asterisk one of my client's office and I have a
very simple setup.

A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a totally digital
setup, it means I have no analogic cards connected.

I can make calls between my extension perfectly;
I can make outgoing calls without any problems;
Incoming calls are dropped after exatly 10 seconds; All incoming calls.

The asterisk box is hooked up to the LAN switch and it runs with a private
IP address. I have another Linux box performing firewall/routing roles.

Outgoing and incoming calls working perfectly from the ATA (linksys pap2t)
but not from asterisk, because it hangs up after 10 seconds.

Some LOGS:

[Mar 16 15:11:12] DEBUG[13311] acl.c: ##### Testing 192.168.20.113 with
192.168.20.0
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 0: OPTIONS
sip:241 at 192.168.20.113:15956;rinstance=542e2865b2c6abe1 SIP/2.0 (71)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 1: Via: SIP/2.0/UDP
192.168.20.249:5060;branch=z9hG4bK29f64fe1;rport (65)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 2: From: "asterisk" <
sip:asterisk at 192.168.20.249 <sip%3Aasterisk at 192.168.20.249>>;tag=as4bdc3589
(61)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 3: To:
<sip:241 at 192.168.20.113:15956;rinstance=542e2865b2c6abe1> (61)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 4: Contact: <
sip:asterisk at 192.168.20.249 <sip%3Aasterisk at 192.168.20.249>> (38)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID:
7a4676c71af6501909db830431000932 at 192.168.20.249 (56)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 7: User-Agent: Asterisk
PBX (24)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 8: Max-Forwards: 70 (16)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 9: Date: Tue, 16 Mar 2010
18:11:12 GMT (35)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 10: Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 11: Supported: replaces
(19)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 12: Content-Length: 0 (17)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: *** SIP TIMER: Initializing
retransmit timer on packet: Id  #-1
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 0: SIP/2.0 200 OK (14)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 1: Via: SIP/2.0/UDP
192.168.20.249:5060;branch=z9hG4bK29f64fe1;rport=5060 (70)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 2: Contact: <sip:
192.168.20.113:15956> (35)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 3: To:
<sip:241 at 192.168.20.113:15956;rinstance=542e2865b2c6abe1>;tag=67747e4a (74)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 4: From: "asterisk"<
sip:asterisk at 192.168.20.249 <sip%3Aasterisk at 192.168.20.249>>;tag=as4bdc3589
(60)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID:
7a4676c71af6501909db830431000932 at 192.168.20.249 (56)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 7: Accept: application/sdp
(23)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 8: Accept-Language: en
(19)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 9: Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 10: User-Agent: X-Lite
release 1104o stamp 56125 (44)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 11: Content-Length: 0 (17)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 12:  (0)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: = Found Their Call ID:
7a4676c71af6501909db830431000932 at 192.168.20.249 Their Tag  Our tag:
as4bdc3589
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: ** SIP TIMER: Cancelling
retransmit of packet (reply received) Retransid #8282
*[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Stopping retransmission on '
7a4676c71af6501909db830431000932 at 192.168.20.249' of Request 102: Match Found
[Mar 16 15:11:13] NOTICE[14413] rtp.c: Unknown RTP codec 126 received from
'192.168.20.113'
[Mar 16 15:11:13] WARNING[13311] chan_sip.c: Maximum retries exceeded on
transmission 22021ea032130d5f3bd50ac67cf61e09 at 200.229.195.226 for seqno 102
(Critical Response)
[Mar 16 15:11:13] DEBUG[13311] chan_sip.c: Setting SIP_ALREADYGONE on dialog
22021ea032130d5f3bd50ac67cf61e09 at 200.229.195.226
[Mar 16 15:11:13] WARNING[13311] chan_sip.c: Hanging up call
22021ea032130d5f3bd50ac67cf61e09 at 200.229.195.226 - no reply to our critical
packet.
[Mar 16 15:11:13] DEBUG[14413] channel.c: Didn't get a frame from channel:
SIP/7977529-081d60d0
*[Mar 16 15:11:13] DEBUG[14413] channel.c: Bridge stops bridging channels
SIP/7977529-081d60d0 and SIP/241-081d7a50
[Mar 16 15:11:13] DEBUG[14413] channel.c: Hanging up channel
'SIP/241-081d7a50'
[Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Hangup call SIP/241-081d7a50, SIP
callid 29d72fed0b17b16b76b12758136b3c25 at 192.168.20.249)
[Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Strict routing enforced for
session 29d72fed0b17b16b76b12758136b3c25 at 192.168.20.249
[Mar 16 15:11:13] DEBUG[14413] chan_sip.c: *** SIP TIMER: Initializing
retransmit timer on packet: Id  #-1
[Mar 16 15:11:13] DEBUG[14413] devicestate.c: Notification of state change
to be queued on device/channel SIP/241-081d7a50
[Mar 16 15:11:13] DEBUG[14413] devicestate.c: Notification of state change
to be queued on device/channel SIP/241
[Mar 16 15:11:13] DEBUG[13304] devicestate.c: No provider found, checking
channel drivers for SIP - 241-081d7a50
[Mar 16 15:11:13] DEBUG[14413] rtp.c: Channel '<unspecified>' has no RTP,
not doing anything
[Mar 16 15:11:13] DEBUG[14413] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[Mar 16 15:11:13] DEBUG[13304] chan_sip.c: Checking device state for peer
241-081d7a50
[Mar 16 15:11:13] DEBUG[14413] pbx.c: Spawn extension
(incoming_calls,7977529,2) exited non-zero on 'SIP/7977529-081d60d0'
[Mar 16 15:11:13] DEBUG[14413] channel.c: Soft-Hanging up channel
'SIP/7977529-081d60d0'
[Mar 16 15:11:13] DEBUG[14413] channel.c: Hanging up channel
'SIP/7977529-081d60d0'
[Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Hangup call SIP/7977529-081d60d0,
SIP callid 22021ea032130d5f3bd50ac67cf61e09 at 200.229.195.226)
[Mar 16 15:11:13] DEBUG[14413] devicestate.c: Notification of state change
to be queued on device/channel SIP/7977529-081d60d0
[Mar 16 15:11:13] DEBUG[14413] devicestate.c: Notification of state change
to be queued on device/channel SIP/7977529

#########################################

And now my extensions.conf and sip.conf

[general]
allowoverlap=no
allowguest=no
bindport=5060
bindaddr=0.0.0.0
externip=189.38.242.109
localnet=192.168.20.0/255.255.255.0
srvlookup=yes
disallow=all
;allow=g729
allow=ulaw
allow=alaw
tos_sip=cs3
tos_audio=ef
tos_video=af41
regcontext=incoming_calls
register=> 7977529 at sip.tellfree.net:PASSWD:7977529 at sip.tellfree.net/7977529

[tellfree]
type=friend
context=incoming_calls
host=sip.tellfree.net
username=7977529
authuser=7977529
authname=7977529
secret=PASSWD
Fromdomain=sip.tellfree.net
fromuser=7977529
insecure=port,invite
qualify=yes
nat=yes
canreinvite=no

[xlite](!)
type=friend
host=dynamic
qualify=yes
context=phones
canreinvite=yes

[241](xlite)
username=241
callerid=241
secret=PASSWD_1

[242](xlite)
username=242
callerid=242
secret=PASSWD_2

[243](xlite)
username=243
callerid=243
secret=PASSWD_3

#############################################

[general]
autofallthrough=yes

[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

[incoming_calls]
;exten => 7977529,1,NoOp()
;exten => 7977529,n,Dial(SIP/241|SIP/243,30,Tt)
exten => 7977529,1,Dial(SIP/241&SIP/243,30,Tt)
;exten => 7977529,n,Dial(SIP/243,30,Tt)
exten => 7977529,n,Hangup()

[outgoing_calls]
exten => _0X.,1,NoOp()
exten => _0X.,n,Dial(Sip/${EXTEN}@tellfree,30,Tt)
exten => _0X.,n,Hangup
exten => _7X.,1,NoOp()
exten => _7X.,n,Dial(Sip/${EXTEN}@tellfree,30,Tt)
exten => _7X.,n,Hangup

[internal]
exten => _24[1-9],1,Verbose(1|Estension ${EXTEN})
exten => _24[1-9],n,SayDigits(${EXTEN})
exten => _24[1-9],n,Dial(SIP/${EXTEN},20,r)
exten => _24[1-9],n,Hangup

[phones]
include => internal
include => outgoing_calls
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