Bruno Camargo
2010-Mar-16 19:55 UTC
[asterisk-users] Asterisk hangup all incoming calls after 10 seconds
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20100316/41150135/attachment.htm
Giorgio Incantalupo
2010-Mar-17 10:04 UTC
[asterisk-users] Asterisk hangup all incoming calls after 10 seconds
Hi Bruno, I remember one of our customer had a similar problem with tellfree in Brazil. Their IT technician told me it was due to a g729 codec problem...they installed it and the problem disappeared. I never checked, I could only trust their man. Maybe it can help. Giorgio P.S.: let me know if it works, please! Bruno Camargo wrote:> 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 > <mailto: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 <mailto: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 > <mailto: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 <http://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 > <mailto: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 > <mailto: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 > <mailto: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 > <mailto: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 > <mailto: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 > <mailto: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 > <mailto: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 > <mailto: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 > <mailto: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 > <mailto: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 <http://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 > <http://ASSWD:7977529 at sip.tellfree.net/7977529> > > [tellfree] > type=friend > context=incoming_calls > host=sip.tellfree.net <http://sip.tellfree.net> > username=7977529 > authuser=7977529 > authname=7977529 > secret=PASSWD > Fromdomain=sip.tellfree.net <http://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