I am doing a little application to originate a call through Asterisk via AMI (Perl Asterisk::Manager). <br>It logs in successfully, does an originate command with<br>Exten: 0020 (which is set up to answer and wait for 60 then hang up)<br>
Channel: SIP/5101234567@test-host (which comes to my desktop machine also running Asterisk).<br><br>At the target machine I see only a CANCEL to which it immediately responds with a No Transaction. Except for every nth try, when I see an INVITE; but only that often.<br>
<br>It
looks like AstMan is asking for a Slin-format connection and the channel is
set up only for Slin or Ulaw but it says "joint capabilities 0x0". Don't know if that's a red herring.<br>
Any advice welcome.<br><br>SIP debug follows:<br><br>[Jul 12 19:08:58] DEBUG[11552] manager.c: Manager received command 'Challenge'<br>[Jul 12 19:08:58] DEBUG[11552] manager.c: Manager received command 'Login'<br>
[Jul 12 19:08:58] DEBUG[11552] config.c: Parsing /etc/asterisk/manager.conf<br>[Jul 12 19:08:58] DEBUG[11552] acl.c: <a href="http://0.0.0.0/0.0.0.0/0.0.0.0">0.0.0.0/0.0.0.0/0.0.0.0</a> appended to acl for peer<br>[Jul 12 19:08:58] DEBUG[11552] acl.c: <a href="http://127.0.0.1/255.255.255.255/255.255.255.255">127.0.0.1/255.255.255.255/255.255.255.255</a> appended to acl for peer<br>
[Jul 12 19:08:58] DEBUG[11552] acl.c: ##### Testing 127.0.0.1 with 0.0.0.0<br>[Jul 12 19:08:58] DEBUG[11552] acl.c: ##### Testing 127.0.0.1 with 127.0.0.1<br>[Jul 12 19:08:58] DEBUG[11552] manager.c: Manager received command 'Originate'<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Asked to create a SIP channel with formats: 0x40 (slin)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Setting NAT on RTP to Off<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** Our native formats are 0x4 (ulaw)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** Joint capabilities are 0x0 (nothing)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** Our capabilities are 0x44 (ulaw|slin)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** Our preferred formats from the incoming channel are 0x40 (slin)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: This channel will not be able to handle video.<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Outgoing Call for 5101234567<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Updating call counter for outgoing call<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Our T38 capability (0), joint T38 capability (0)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: ** Our capability: 0x44 (ulaw|slin) Video flag: False<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: ** Our prefcodec: 0x40 (slin)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: -- Done with adding codecs to SDP<br>
[Jul 12 19:08:58] DEBUG[11552] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Done building SDP. Settling with this capability: 0x44 (ulaw|slin)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 0: INVITE <a href="mailto:sip%3A5101234567@209.204.152.219">sip:5101234567@209.204.152.219</a> SIP/2.0 (45)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.253.35:5060;branch=z9hG4bK23a50508;rport (65)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 2: From: "asterisk" <<a href="mailto:sip%3Aasterisk@192.168.253.35">sip:asterisk@192.168.253.35</a>>;tag=as1826f152 (61)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 3: To: <<a href="mailto:sip%3A5101234567@209.204.152.219">sip:5101234567@209.204.152.219</a>> (36)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 4: Contact: <<a href="mailto:sip%3Aasterisk@192.168.253.35">sip:asterisk@192.168.253.35</a>> (38)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 5: Call-ID: <a href="mailto:16ba645535fc8f7157940d6226192689@192.168.253.35">16ba645535fc8f7157940d6226192689@192.168.253.35</a> (56)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 6: CSeq: 102 INVITE (16)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 8: Max-Forwards: 70 (16)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 9: Date: Mon, 13 Jul 2009 02:08:58 GMT (35)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 11: Supported: replaces (19)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 12: Content-Type: application/sdp (29)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 13: Content-Length: 213 (19)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Header 14: (0)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Line: v=0 (3)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Line: o=root 11287 11287 IN IP4 192.168.253.35 (40)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Line: s=session (9)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Line: c=IN IP4 192.168.253.35 (23)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Line: t=0 0 (5)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Line: m=audio 15220 RTP/AVP 10 0 (26)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Line: a=rtpmap:10 L16/8000 (20)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Line: a=silenceSupp:off - - - - (25)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Line: a=ptime:20 (10)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Line: a=sendrecv (10)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1<br>
[Jul 12 19:08:58] DEBUG[11552] channel.c: Hanging up channel 'SIP/sip-flat5th-081d05e8'<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Hangup call SIP/sip-flat5th-081d05e8, SIP callid <a href="mailto:16ba645535fc8f7157940d6226192689@192.168.253.35">16ba645535fc8f7157940d6226192689@192.168.253.35</a>)<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Hanging up channel in state Down (not UP)<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Acked pending invite 102<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #36<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Stopping retransmission on '<a href="mailto:16ba645535fc8f7157940d6226192689@192.168.253.35">16ba645535fc8f7157940d6226192689@192.168.253.35</a>' of Request 102: Match Found<br>
[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1<br>[Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Updating call counter for outgoing call<br>[Jul 12 19:08:58] DEBUG[11552] devicestate.c: Notification of state change to be queued on device/channel SIP/sip-flat5th<br>
[Jul 12 19:08:58] DEBUG[11552] devicestate.c: Notification of state change to be queued on device/channel<br>[Jul 12 19:08:58] DEBUG[11290] devicestate.c: No provider found, checking channel drivers for SIP - sip-flat5th<br>
[Jul 12 19:08:58] DEBUG[11290] chan_sip.c: Checking device state for peer sip-flat5th<br>[Jul 12 19:08:58] DEBUG[11290] devicestate.c: Changing state for SIP/sip-flat5th - state 1 (Not in use)<br>[Jul 12 19:08:58] DEBUG[11290] devicestate.c: Checking if I can find provider for "" - number: (null)<br>
[Jul 12 19:08:58] DEBUG[11290] devicestate.c: Checking provider Park with<br>[Jul 12 19:08:58] DEBUG[11290] devicestate.c: Changing state for - state 4 (Invalid)<br>[Jul 12 19:08:58] DEBUG[11314] app_queue.c: Device 'SIP/sip-flat5th' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.<br>
[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: Header 0: SIP/2.0 481 Call leg/transaction does not exist (47)<br>[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: Header 1: Via: SIP/2.0/UDP 209.204.152.219:29132;branch=z9hG4bK23a50508;received=204.86.255.250;rport=29132 (97)<br>
[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: Header 2: From: "asterisk" <<a href="mailto:sip%3Aasterisk@192.168.253.35">sip:asterisk@192.168.253.35</a>>;tag=as1826f152 (61)<br>[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: Header 3: To: <<a href="mailto:sip%3A5101234567@209.204.152.219">sip:5101234567@209.204.152.219</a>>;tag=as79658769 (51)<br>
[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: Header 4: Call-ID: <a href="mailto:16ba645535fc8f7157940d6226192689@192.168.253.35">16ba645535fc8f7157940d6226192689@192.168.253.35</a> (56)<br>[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: Header 5: CSeq: 102 CANCEL (16)<br>
[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24)<br>[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)<br>[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: Header 8: Supported: replaces (19)<br>
[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: Header 9: Content-Length: 0 (17)<br>[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: Header 10: (0)<br>[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: = Found Their Call ID: <a href="mailto:16ba645535fc8f7157940d6226192689@192.168.253.35">16ba645535fc8f7157940d6226192689@192.168.253.35</a> Their Tag Our tag: as1826f152<br>
[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #38<br>[Jul 12 19:08:58] DEBUG[11313] chan_sip.c: Stopping retransmission on '<a href="mailto:16ba645535fc8f7157940d6226192689@192.168.253.35">16ba645535fc8f7157940d6226192689@192.168.253.35</a>' of Request 102: Match Found<br>
[Jul 12 19:09:30] DEBUG[11313] chan_sip.c: Auto destroying SIP dialog '<a href="mailto:16ba645535fc8f7157940d6226192689@192.168.253.35">16ba645535fc8f7157940d6226192689@192.168.253.35</a>'<br>[Jul 12 19:09:30] DEBUG[11313] chan_sip.c: Destroying SIP dialog <a href="mailto:16ba645535fc8f7157940d6226192689@192.168.253.35">16ba645535fc8f7157940d6226192689@192.168.253.35</a><br>
<br>