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 < [email protected]> 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:[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] <sip%[email protected]> > > <mailto:sip%[email protected] > > <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] <sip%[email protected]> <mailto: > sip%[email protected] <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] <sip%[email protected]> > > <mailto:sip%[email protected] > > <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? 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 > -- BrCaBadT
-- _____________________________________________________________________ -- 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
