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

Alexandru Oniciuc Alexandru.Oniciuc at trivenet.it
Wed Mar 17 05:27:41 CDT 2010 

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-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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!

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@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


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

More information about the asterisk-users mailing list