Hello Gentleman,<br><br>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.<br><br>Here's the thing.<br>
<br>I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup.<br><br>A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a totally digital setup, it means I have no analogic cards connected.<br>
<br>I can make calls between my extension perfectly;<br>I can make outgoing calls without any problems;<br>Incoming calls are dropped after exatly 10 seconds; All incoming calls.<br><br>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.<br>
<br>Outgoing and incoming calls working perfectly from the ATA (linksys pap2t) but not from asterisk, because it hangs up after 10 seconds.<br><br>Some LOGS:<br><br>[Mar 16 15:11:12] DEBUG[13311] acl.c: ##### Testing 192.168.20.113 with 192.168.20.0<br>
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 0: OPTIONS sip:241@192.168.20.113:15956;rinstance=542e2865b2c6abe1 SIP/2.0 (71)<br>[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)<br>
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 2: From: "asterisk" <<a href="mailto:sip%3Aasterisk@192.168.20.249">sip:asterisk@192.168.20.249</a>>;tag=as4bdc3589 (61)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 3: To: <sip:241@192.168.20.113:15956;rinstance=542e2865b2c6abe1> (61)<br>
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 4: Contact: <<a href="mailto:sip%3Aasterisk@192.168.20.249">sip:asterisk@192.168.20.249</a>> (38)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID: <a href="mailto:7a4676c71af6501909db830431000932@192.168.20.249">7a4676c71af6501909db830431000932@192.168.20.249</a> (56)<br>
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 8: Max-Forwards: 70 (16)<br>
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 9: Date: Tue, 16 Mar 2010 18:11:12 GMT (35)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)<br>
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 11: Supported: replaces (19)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 12: Content-Length: 0 (17)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1<br>
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 0: SIP/2.0 200 OK (14)<br>[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)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 2: Contact: <sip:<a href="http://192.168.20.113:15956">192.168.20.113:15956</a>> (35)<br>
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 3: To: <sip:241@192.168.20.113:15956;rinstance=542e2865b2c6abe1>;tag=67747e4a (74)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 4: From: "asterisk"<<a href="mailto:sip%3Aasterisk@192.168.20.249">sip:asterisk@192.168.20.249</a>>;tag=as4bdc3589 (60)<br>
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID: <a href="mailto:7a4676c71af6501909db830431000932@192.168.20.249">7a4676c71af6501909db830431000932@192.168.20.249</a> (56)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17)<br>
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 7: Accept: application/sdp (23)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 8: Accept-Language: en (19)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81)<br>
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 10: User-Agent: X-Lite release 1104o stamp 56125 (44)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 11: Content-Length: 0 (17)<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 12: (0)<br>
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: = Found Their Call ID: <a href="mailto:7a4676c71af6501909db830431000932@192.168.20.249">7a4676c71af6501909db830431000932@192.168.20.249</a> Their Tag Our tag: as4bdc3589<br>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8282<br>
<b>[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Stopping retransmission on '<a href="mailto:7a4676c71af6501909db830431000932@192.168.20.249">7a4676c71af6501909db830431000932@192.168.20.249</a>' of Request 102: Match Found<br>
[Mar 16 15:11:13] NOTICE[14413] rtp.c: Unknown RTP codec 126 received from '192.168.20.113'<br>[Mar 16 15:11:13] WARNING[13311] chan_sip.c: Maximum retries exceeded on transmission <a href="mailto:22021ea032130d5f3bd50ac67cf61e09@200.229.195.226">22021ea032130d5f3bd50ac67cf61e09@200.229.195.226</a> for seqno 102 (Critical Response)<br>
[Mar 16 15:11:13] DEBUG[13311] chan_sip.c: Setting SIP_ALREADYGONE on dialog <a href="mailto:22021ea032130d5f3bd50ac67cf61e09@200.229.195.226">22021ea032130d5f3bd50ac67cf61e09@200.229.195.226</a><br>[Mar 16 15:11:13] WARNING[13311] chan_sip.c: Hanging up call <a href="mailto:22021ea032130d5f3bd50ac67cf61e09@200.229.195.226">22021ea032130d5f3bd50ac67cf61e09@200.229.195.226</a> - no reply to our critical packet.<br>
[Mar 16 15:11:13] DEBUG[14413] channel.c: Didn't get a frame from channel: SIP/7977529-081d60d0<br></b>[Mar 16 15:11:13] DEBUG[14413] channel.c: Bridge stops bridging channels SIP/7977529-081d60d0 and SIP/241-081d7a50<br>
[Mar 16 15:11:13] DEBUG[14413] channel.c: Hanging up channel 'SIP/241-081d7a50'<br>[Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Hangup call SIP/241-081d7a50, SIP callid <a href="mailto:29d72fed0b17b16b76b12758136b3c25@192.168.20.249">29d72fed0b17b16b76b12758136b3c25@192.168.20.249</a>)<br>
[Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Strict routing enforced for session <a href="mailto:29d72fed0b17b16b76b12758136b3c25@192.168.20.249">29d72fed0b17b16b76b12758136b3c25@192.168.20.249</a><br>[Mar 16 15:11:13] DEBUG[14413] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1<br>
[Mar 16 15:11:13] DEBUG[14413] devicestate.c: Notification of state change to be queued on device/channel SIP/241-081d7a50<br>[Mar 16 15:11:13] DEBUG[14413] devicestate.c: Notification of state change to be queued on device/channel SIP/241<br>
[Mar 16 15:11:13] DEBUG[13304] devicestate.c: No provider found, checking channel drivers for SIP - 241-081d7a50<br>[Mar 16 15:11:13] DEBUG[14413] rtp.c: Channel '<unspecified>' has no RTP, not doing anything<br>
[Mar 16 15:11:13] DEBUG[14413] app_dial.c: Exiting with DIALSTATUS=ANSWER.<br>[Mar 16 15:11:13] DEBUG[13304] chan_sip.c: Checking device state for peer 241-081d7a50<br>[Mar 16 15:11:13] DEBUG[14413] pbx.c: Spawn extension (incoming_calls,7977529,2) exited non-zero on 'SIP/7977529-081d60d0'<br>
[Mar 16 15:11:13] DEBUG[14413] channel.c: Soft-Hanging up channel 'SIP/7977529-081d60d0'<br>[Mar 16 15:11:13] DEBUG[14413] channel.c: Hanging up channel 'SIP/7977529-081d60d0'<br>[Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Hangup call SIP/7977529-081d60d0, SIP callid <a href="mailto:22021ea032130d5f3bd50ac67cf61e09@200.229.195.226">22021ea032130d5f3bd50ac67cf61e09@200.229.195.226</a>)<br>
[Mar 16 15:11:13] DEBUG[14413] devicestate.c: Notification of state change to be queued on device/channel SIP/7977529-081d60d0<br>[Mar 16 15:11:13] DEBUG[14413] devicestate.c: Notification of state change to be queued on device/channel SIP/7977529<br>
<br>#########################################<br><br>And now my extensions.conf and sip.conf<br><br>[general]<br>allowoverlap=no<br>allowguest=no<br>bindport=5060<br>bindaddr=0.0.0.0<br>externip=189.38.242.109<br>localnet=<a href="http://192.168.20.0/255.255.255.0">192.168.20.0/255.255.255.0</a><br>
srvlookup=yes<br>disallow=all<br>;allow=g729<br>allow=ulaw<br>allow=alaw<br>tos_sip=cs3<br>tos_audio=ef<br>tos_video=af41<br>regcontext=incoming_calls<br>register=> 7977529@sip.tellfree.net:P<a href="http://ASSWD:7977529@sip.tellfree.net/7977529">ASSWD:7977529@sip.tellfree.net/7977529</a><br>
<br>[tellfree]<br>type=friend<br>context=incoming_calls<br>host=<a href="http://sip.tellfree.net">sip.tellfree.net</a><br>username=7977529<br>authuser=7977529<br>authname=7977529<br>secret=PASSWD<br>Fromdomain=<a href="http://sip.tellfree.net">sip.tellfree.net</a><br>
fromuser=7977529<br>insecure=port,invite<br>qualify=yes<br>nat=yes<br>canreinvite=no<br><br>[xlite](!)<br>type=friend<br>host=dynamic<br>qualify=yes<br>context=phones<br>canreinvite=yes<br><br>[241](xlite)<br>username=241<br>
callerid=241<br>secret=PASSWD_1<br><br>[242](xlite)<br>username=242<br>callerid=242<br>secret=PASSWD_2<br><br>[243](xlite)<br>username=243<br>callerid=243<br>secret=PASSWD_3<br><br>#############################################<br>
<br>[general]<br>autofallthrough=yes<br><br>[default]<br>exten => s,1,Verbose(1|Unrouted call handler)<br>exten => s,n,Answer()<br>exten => s,n,Wait(1)<br>exten => s,n,Playback(tt-weasels)<br>exten => s,n,Hangup()<br>
<br>[incoming_calls]<br>;exten => 7977529,1,NoOp()<br>;exten => 7977529,n,Dial(SIP/241|SIP/243,30,Tt)<br>exten => 7977529,1,Dial(SIP/241&SIP/243,30,Tt)<br>;exten => 7977529,n,Dial(SIP/243,30,Tt)<br>exten => 7977529,n,Hangup()<br>
<br>[outgoing_calls]<br>exten => _0X.,1,NoOp()<br>exten => _0X.,n,Dial(Sip/${EXTEN}@tellfree,30,Tt)<br>exten => _0X.,n,Hangup<br>exten => _7X.,1,NoOp()<br>exten => _7X.,n,Dial(Sip/${EXTEN}@tellfree,30,Tt)<br>
exten => _7X.,n,Hangup<br><br>[internal]<br>exten => _24[1-9],1,Verbose(1|Estension ${EXTEN})<br>exten => _24[1-9],n,SayDigits(${EXTEN})<br>exten => _24[1-9],n,Dial(SIP/${EXTEN},20,r)<br>exten => _24[1-9],n,Hangup<br>
<br>[phones]<br>include => internal<br>include => outgoing_calls<br>