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]>>;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]>> (38) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID: [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> (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]>>;tag=as4bdc3589 (60) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID: [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] 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]' 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] for seqno 102 (Critical Response) [Mar 16 15:11:13] DEBUG[13311] chan_sip.c: Setting SIP_ALREADYGONE on dialog [email protected] [Mar 16 15:11:13] WARNING[13311] chan_sip.c: Hanging up call [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]) [Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Strict routing enforced for session [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]) [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=> [email protected]:PASSWD:[email protected]/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/${ext...@tellfree,30,Tt) exten => _0X.,n,Hangup exten => _7X.,1,NoOp() exten => _7X.,n,Dial(Sip/${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
