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

Discussion:

Bruno Camargo

2010-03-16 19:55:18 UTC

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:***@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:***@192.168.20.249 <sip%***@192.168.20.249>>;tag=as4bdc3589
(61)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 3: To:
<sip:***@192.168.20.113:15956;rinstance=542e2865b2c6abe1> (61)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 4: Contact: <
sip:***@192.168.20.249 <sip%***@192.168.20.249>> (38)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID:
***@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:***@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:***@192.168.20.249 <sip%***@192.168.20.249>>;tag=as4bdc3589
(60)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID:
***@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:
***@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 '
***@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 ***@200.229.195.226 for seqno 102
(Critical Response)
[Mar 16 15:11:13] DEBUG[13311] chan_sip.c: Setting SIP_ALREADYGONE on dialog
***@200.229.195.226
[Mar 16 15:11:13] WARNING[13311] chan_sip.c: Hanging up call
***@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 ***@192.168.20.249)
[Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Strict routing enforced for
session ***@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 ***@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=> ***@sip.tellfree.net:PASSWD:***@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

Giorgio Incantalupo

2010-03-17 10:04:27 UTC

Permalink

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!

Post by Bruno Camargo
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.
[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
[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"
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17)
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)
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)
<sip:192.168.20.113:15956 <http://192.168.20.113:15956>> (35)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17)
application/sdp (23)
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)
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)
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
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
102 (Critical Response)
[Mar 16 15:11:13] DEBUG[13311] chan_sip.c: Setting SIP_ALREADYGONE on
[Mar 16 15:11:13] WARNING[13311] chan_sip.c: Hanging up call
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
[Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Strict routing enforced for
[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
[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=>
[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,Hangup
exten => _7X.,1,NoOp()
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

Alexandru Oniciuc

2010-03-17 10:27:41 UTC

Permalink

NOTICE[14413] rtp.c: Unknown RTP codec 126 received from '192.168.20.113'

Maybe the codec 126 is the problem?

[core] show codecs
[core] show translation

Maybe you don't have the codec required by your provider.

Regards,
Alex

-----Messaggio originale-----
Da: asterisk-users-***@lists.digium.com [mailto:asterisk-users-***@lists.digium.com] Per conto di Giorgio Incantalupo
Inviato: mercoledì 17 marzo 2010 11:04
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [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!

--
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Bruno Camargo

2010-03-17 17:05:56 UTC

Permalink

Hi Giorgio,

So it means that Asterisk has no native support for g729 ?

Thanks

On Wed, Mar 17, 2010 at 7:04 AM, Giorgio Incantalupo <

Post by Giorgio Incantalupo
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!

(61)

Post by Bruno Camargo
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17)
Asterisk PBX (24)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 8: Max-Forwards: 70

(16)

Post by Bruno Camargo
[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)
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)
<sip:192.168.20.113:15956 <http://192.168.20.113:15956>> (35)

(60)

Post by Bruno Camargo
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17)
application/sdp (23)
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)
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)
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
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
102 (Critical Response)
[Mar 16 15:11:13] DEBUG[13311] chan_sip.c: Setting SIP_ALREADYGONE on
[Mar 16 15:11:13] WARNING[13311] chan_sip.c: Hanging up call
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
[Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Strict routing enforced for
[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.

Post by Bruno Camargo
[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
[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=>
[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,Hangup
exten => _7X.,1,NoOp()
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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Fred Posner

2010-03-17 17:12:58 UTC

Permalink

Post by Bruno Camargo
Hi Giorgio,
So it means that Asterisk has no native support for g729 ?
Thanks
--
BrCaBadT
--

Bruno Camargo

2010-03-18 01:05:43 UTC

Permalink

Hello Gentleman,

I guess the problem was the codec.

I have allowed only g711u for testing purposes and the incoming call endured
for 1 minute, until the caller hanged.

Thanks a lot for the support.... but there are tons of questions yet to be
answered!

Thanks

Post by Fred Posner

Post by Bruno Camargo
Hi Giorgio,
So it means that Asterisk has no native support for g729 ?
Thanks
--
BrCaBadT
--

Depends on your definition of support. It supports passthrough... but if
you're using it locally on a bridge on transcoding, you'll need to purchase
a license. The codec itself is non-G729 compliant.
---fred
http://qxork.com
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asterisk-users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

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Bruno Camargo

2010-03-18 01:30:38 UTC

Permalink

Toooooo early... call droped after 11 seconds now... different log.

[Mar 17 22:19:17] DEBUG[2783] chan_sip.c: SIP TIMER: Rescheduling
retransmission #13781 (6) SIP/2.0 - 1
[Mar 17 22:19:17] DEBUG[2783] chan_sip.c: ** SIP timers: Rescheduling
retransmission 7 to 4000 ms (t1 100 ms (Retrans id #13781))
[Mar 17 22:19:20] DEBUG[4459] rtp.c: Got RTCP report of 176 bytes
[Mar 17 22:19:21] WARNING[2783] chan_sip.c: Maximum retries exceeded on
transmission ***@200.229.195.226 for seqno 102
(Critical Response)
[Mar 17 22:19:21] DEBUG[2783] chan_sip.c: Setting SIP_ALREADYGONE on dialog
***@200.229.195.226
[Mar 17 22:19:21] WARNING[2783] chan_sip.c: Hanging up call
***@200.229.195.226 - no reply to our critical
packet.
[Mar 17 22:19:21] DEBUG[4459] channel.c: Didn't get a frame from channel:
SIP/7977529-081931c0
[Mar 17 22:19:21] DEBUG[4459] channel.c: Bridge stops bridging channels
SIP/7977529-081931c0 and SIP/242-081910e8
[Mar 17 22:19:21] DEBUG[4459] channel.c: Hanging up channel
'SIP/242-081910e8'
[Mar 17 22:19:21] DEBUG[4459] chan_sip.c: Hangup call SIP/242-081910e8, SIP
callid ***@192.168.20.249)
[Mar 17 22:19:21] DEBUG[4459] chan_sip.c: Strict routing enforced for
session ***@192.168.20.249
[Mar 17 22:19:21] DEBUG[4459] chan_sip.c: *** SIP TIMER: Initializing
retransmit timer on packet: Id #-1

and not a clue.

THanks alot

Post by Bruno Camargo
Hello Gentleman,
I guess the problem was the codec.
I have allowed only g711u for testing purposes and the incoming call
endured for 1 minute, until the caller hanged.
Thanks a lot for the support.... but there are tons of questions yet to be
answered!
Thanks

Post by Fred Posner

Post by Bruno Camargo
Hi Giorgio,
So it means that Asterisk has no native support for g729 ?
Thanks
--
BrCaBadT
--

--
BrCaBadT