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

Wed Mar 17 12:05:56 CDT 2010

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 <
gincantalupo at fgasoftware.com> wrote:

> 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 <sip%3Aasterisk at 192.168.20.249>
> > <mailto:sip%3Aasterisk at 192.168.20.249 <sip%253Aasterisk 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> <mailto:
> sip%3Aasterisk at 192.168.20.249 <sip%253Aasterisk 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 <sip%3Aasterisk at 192.168.20.249>
> > <mailto:sip%3Aasterisk at 192.168.20.249 <sip%253Aasterisk 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@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
>
>
> --
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-- 
BrCaBadT
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