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:[email protected]: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:[email protected] > <mailto:sip%[email protected]>>;tag=as4bdc3589 (61) > [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 3: To: > <sip:[email protected]:15956;rinstance=542e2865b2c6abe1> (61) > [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 4: Contact: > <sip:[email protected] <mailto:sip%[email protected]>> (38) > [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID: > [email protected] > <mailto:[email protected]> (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:[email protected]:15956;rinstance=542e2865b2c6abe1>;tag=67747e4a > (74) > [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 4: From: > "asterisk"<sip:[email protected] > <mailto:sip%[email protected]>>;tag=as4bdc3589 (60) > [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID: > [email protected] > <mailto:[email protected]> (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: > [email protected] > <mailto:[email protected]> 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 > '[email protected] > <mailto:[email protected]>' 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 [email protected] > <mailto:[email protected]> for seqno > 102 (Critical Response) > [Mar 16 15:11:13] DEBUG[13311] chan_sip.c: Setting SIP_ALREADYGONE on > dialog [email protected] > <mailto:[email protected]> > [Mar 16 15:11:13] WARNING[13311] chan_sip.c: Hanging up call > [email protected] > <mailto:[email protected]> - 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 > [email protected] > <mailto:[email protected]>) > [Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Strict routing enforced for > session [email protected] > <mailto:[email protected]> > [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 > [email protected] > <mailto:[email protected]>) > [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=> > [email protected]:PASSWD:[email protected]/7977529 > <http://ASSWD:[email protected]/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/${ext...@tellfree,30,Tt) > exten => _0X.,n,Hangup > exten => _7X.,1,NoOp() > exten => _7X.,n,Dial(Sip/${ext...@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? 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