Hi
I've spent two days trying to solve this issue but to no prevail and I'm
hoping to get some help.
I've configured Asterisk as a SIP client, running on OpenWRT on an embedded
device with onboard FXS and ATA. Asterisk is connecting to an external SIP
provider on the Internet who in turn provides a PSTN gateway. I'm able to
make calls to other SIP accounts registered on the same server who are
outside my LAN. However, I can not make calls to any PSTN numbers. When
trying to make PSTN calls it sounds like the person at the other end is
immediately rejecting the call although I know this is not the case.
Firstly, I'm absolutely sure that the PSTN gateway is working because I can
make outbound PSTN calls with the same SIP account using other SIP clients
(Empathy-SIP, SIPDroid) from the same LAN. However, when registering the
same SIP account using Asterisk from OpenWRT all PSTN calls fail. Inbound
calls from PSTN numbers also fail while calls from other SIP clients on the
same server work fine. Thus, I'm fairly confident the problem is with my
Asterisk configuration.
The SIP accounts shows as registered in Asterisk. I've attached detailed
error logs. The log files 'messages-pstn.log' shows the failed (PSTN) call
and 'messages-voip.log' shows the successful (VOIP) call. Note that I have
replaced actual phone numbers and domain names with *** for anonymity.
I suspect perhaps a codec issue, but I haven't been able to identify the
actual problem. Any ideas that will help me towards solving this problem is
greatly appreciated.
Regards,
Helge
[Feb 10 16:40:56] VERBOSE[5769] logger.c: -- event_offhook
[Feb 10 16:40:56] VERBOSE[5769] logger.c: -- AST_STATE_DOWN:
[Feb 10 16:40:56] VERBOSE[5769] logger.c: -- start mp_new
[Feb 10 16:40:59] VERBOSE[5769] logger.c: -- event_dtmf #
[Feb 10 16:40:59] DEBUG[5769] devicestate.c: Notification of state change to be queued on device/channel MP/1
[Feb 10 16:40:59] DEBUG[5767] devicestate.c: Changing state for MP/1 - state 0 (Unknown)
[Feb 10 16:40:59] VERBOSE[5769] logger.c: -- event_dtmf *
[Feb 10 16:40:59] VERBOSE[5769] logger.c: -- event_dtmf *
[Feb 10 16:41:00] VERBOSE[5769] logger.c: -- event_dtmf *
[Feb 10 16:41:00] VERBOSE[5769] logger.c: -- event_dtmf *
[Feb 10 16:41:00] VERBOSE[5769] logger.c: -- event_dtmf *
[Feb 10 16:41:00] VERBOSE[5769] logger.c: -- event_dtmf *
[Feb 10 16:41:00] VERBOSE[5769] logger.c: -- event_dtmf *
[Feb 10 16:41:01] VERBOSE[5769] logger.c: -- event_dtmf *
[Feb 10 16:41:01] VERBOSE[5769] logger.c: -- event_dtmf *
[Feb 10 16:41:01] VERBOSE[5769] logger.c: -- event_dtmf *
[Feb 10 16:41:04] VERBOSE[5769] logger.c: -- event_digit_timer
[Feb 10 16:41:04] VERBOSE[5769] logger.c: -- extension exists, starting PBX #**********
[Feb 10 16:41:04] DEBUG[5769] devicestate.c: Notification of state change to be queued on device/channel MP/1
[Feb 10 16:41:04] DEBUG[5767] devicestate.c: Changing state for MP/1 - state 0 (Unknown)
[Feb 10 16:41:04] DEBUG[5901] pbx.c: Launching 'Dial'
[Feb 10 16:41:04] VERBOSE[5901] logger.c: -- Executing [#**********@default:1] Dial("MP/1", "SIP/**********@sipaccount|120|r") in new stack
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw)
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Setting NAT on RTP to On
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: *** Our native formats are 0x2 (gsm)
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: *** Joint capabilities are 0x0 (nothing)
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: *** Our capabilities are 0xe (gsm|ulaw|alaw)
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x2 (gsm)
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw)
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: This channel will not be able to handle video.
[Feb 10 16:41:04] DEBUG[5901] rtp.c: Channel 'MP/1' has no RTP, not doing anything
[Feb 10 16:41:04] DEBUG[5901] channel.c: Not copying variable STACK-default-#**********-1.
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Outgoing Call for **********
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Updating call counter for outgoing call
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Our T38 capability (0), joint T38 capability (0)
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: ** Our capability: 0x6 (gsm|ulaw) Video flag: False
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
[Feb 10 16:41:04] VERBOSE[5901] logger.c: Audio is at 10.130.1.21 port 17800
[Feb 10 16:41:04] VERBOSE[5901] logger.c: Adding codec 0x4 (ulaw) to SDP
[Feb 10 16:41:04] VERBOSE[5901] logger.c: Adding codec 0x2 (gsm) to SDP
[Feb 10 16:41:04] VERBOSE[5901] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: -- Done with adding codecs to SDP
[Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Done building SDP. Settling with this capability: 0x6 (gsm|ulaw)
[Feb 10 16:41:04] VERBOSE[5901] logger.c: Reliably Transmitting (NAT) to 66.8.50.218:5060:
INVITE sip:**********@sip.*****.co.za SIP/2.0
Via: SIP/2.0/UDP 10.130.1.21:5060;branch=z9hG4bK7565f7f2;rport
From: "asterisk" <sip:**********@10.130.1.21>;tag=as5771f244
To: <sip:**********@sip.*****.co.za>
Contact: <sip:**********@10.130.1.21>
Call-ID: 50ad28aa755e4506449616ce1f766fa1@10.130.1.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 10 Feb 2011 16:41:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 5901 5901 IN IP4 10.130.1.21
s=session
c=IN IP4 10.130.1.21
t=0 0
m=audio 17800 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Feb 10 16:41:04] DEBUG[5901] sched.c: Attempted to delete nonexistent schedule entry 0!
[Feb 10 16:41:04] VERBOSE[5901] logger.c: -- Called **********@sipaccount
[Feb 10 16:41:04] VERBOSE[5901] logger.c: -- Asked to indicate 'Remote end is ringing' condition on channel MP/1
[Feb 10 16:41:05] VERBOSE[5770] logger.c:
<--- SIP read from 66.8.50.218:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.130.1.21:5060;branch=z9hG4bK7565f7f2;rport=5060
From: "asterisk" <sip:**********@10.130.1.21:5060>;tag=as5771f244
To: <sip:**********@sip.*****.co.za>;tag=4a858b1e222aafec6644321f187dd12e.c22d
Call-ID: 50ad28aa755e4506449616ce1f766fa1@10.130.1.21
CSeq: 102 INVITE
Server: OpenSIPS (1.4.2-notls (i386/linux))
Content-Length: 0
<------------->
[Feb 10 16:41:05] VERBOSE[5770] logger.c: --- (8 headers 0 lines) ---
[Feb 10 16:41:05] DEBUG[5770] chan_sip.c: Acked pending invite 102
[Feb 10 16:41:05] DEBUG[5770] chan_sip.c: Stopping retransmission on '50ad28aa755e4506449616ce1f766fa1@10.130.1.21' of Request 102: Match Not Found
[Feb 10 16:41:05] VERBOSE[5770] logger.c: Transmitting (NAT) to 66.8.50.218:5060:
ACK sip:**********@sip.*****.co.za SIP/2.0
Via: SIP/2.0/UDP 10.130.1.21:5060;branch=z9hG4bK7565f7f2;rport
From: "asterisk" <sip:**********@10.130.1.21>;tag=as5771f244
To: <sip:**********@sip.*****.co.za>;tag=4a858b1e222aafec6644321f187dd12e.c22d
Contact: <sip:**********@10.130.1.21>
Call-ID: 50ad28aa755e4506449616ce1f766fa1@10.130.1.21
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
[Feb 10 16:41:05] DEBUG[5770] chan_sip.c: Setting SIP_ALREADYGONE on dialog 50ad28aa755e4506449616ce1f766fa1@10.130.1.21
[Feb 10 16:41:05] VERBOSE[5901] logger.c: -- SIP/sipaccount-00567370 is circuit-busy
[Feb 10 16:41:05] DEBUG[5901] channel.c: Hanging up channel 'SIP/sipaccount-00567370'
[Feb 10 16:41:05] DEBUG[5901] chan_sip.c: Hangup call SIP/sipaccount-00567370, SIP callid 50ad28aa755e4506449616ce1f766fa1@10.130.1.21)
[Feb 10 16:41:05] DEBUG[5901] devicestate.c: Notification of state change to be queued on device/channel SIP/sipaccount-00567370
[Feb 10 16:41:05] DEBUG[5767] chan_sip.c: Checking device state for peer sipaccount
[Feb 10 16:41:05] DEBUG[5767] devicestate.c: Changing state for SIP/sipaccount - state 1 (Not in use)
[Feb 10 16:41:05] VERBOSE[5901] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
[Feb 10 16:41:05] VERBOSE[5901] logger.c: -- Asked to indicate 'Stop tone' condition on channel MP/1
[Feb 10 16:41:05] DEBUG[5901] rtp.c: Channel 'MP/1' has no RTP, not doing anything
[Feb 10 16:41:05] DEBUG[5901] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
[Feb 10 16:41:05] VERBOSE[5901] logger.c: == Auto fallthrough, channel 'MP/1' status is 'CONGESTION'
[Feb 10 16:41:05] VERBOSE[5901] logger.c: -- Asked to indicate 'Congestion (circuits busy)' condition on channel MP/1
[Feb 10 16:41:05] DEBUG[5901] devicestate.c: Notification of state change to be queued on device/channel MP/1
[Feb 10 16:41:05] DEBUG[5767] devicestate.c: Changing state for MP/1 - state 0 (Unknown)
[Feb 10 16:41:05] DEBUG[5770] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
[Feb 10 16:41:05] VERBOSE[5770] logger.c: Reliably Transmitting (NAT) to 66.8.50.218:5060:
OPTIONS sip:sip.*****.co.za SIP/2.0
Via: SIP/2.0/UDP 10.130.1.21:5060;branch=z9hG4bK0453341e;rport
From: "asterisk" <sip:asterisk@10.130.1.21>;tag=as4668587d
To: <sip:sip.*****.co.za>
Contact: <sip:asterisk@10.130.1.21>
Call-ID: 627ecef11dd58a8d49069b7b42166f5b@10.130.1.21
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 10 Feb 2011 16:41:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Feb 10 16:41:05] DEBUG[5770] sched.c: Attempted to delete nonexistent schedule entry 0!
[Feb 10 16:41:05] VERBOSE[5770] logger.c: Really destroying SIP dialog '50ad28aa755e4506449616ce1f766fa1@10.130.1.21' Method: INVITE
[Feb 10 16:41:05] VERBOSE[5770] logger.c:
<--- SIP read from 66.8.50.218:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.130.1.21:5060;branch=z9hG4bK0453341e;rport=5060
From: "asterisk" <sip:asterisk@10.130.1.21:5060>;tag=as4668587d
To: <sip:sip.*****.co.za>;tag=4a858b1e222aafec6644321f187dd12e.2fa0
Call-ID: 627ecef11dd58a8d49069b7b42166f5b@10.130.1.21
CSeq: 102 OPTIONS
Server: OpenSIPS (1.4.2-notls (i386/linux))
Content-Length: 0
<------------->
[Feb 10 16:41:05] VERBOSE[5770] logger.c: --- (8 headers 0 lines) ---
[Feb 10 16:41:05] DEBUG[5770] chan_sip.c: Stopping retransmission on '627ecef11dd58a8d49069b7b42166f5b@10.130.1.21' of Request 102: Match Not Found
[Feb 10 16:41:05] VERBOSE[5770] logger.c: Really destroying SIP dialog '627ecef11dd58a8d49069b7b42166f5b@10.130.1.21' Method: OPTIONS
[Feb 10 16:41:07] VERBOSE[5769] logger.c: -- event_onhook
[Feb 10 16:41:07] VERBOSE[5769] logger.c: -- default: hangup sound_on = 1
[Feb 10 16:41:07] DEBUG[5901] channel.c: Soft-Hanging up channel 'MP/1'
[Feb 10 16:41:07] DEBUG[5901] channel.c: Hanging up channel 'MP/1'
[Feb 10 16:41:07] VERBOSE[5901] logger.c: -- start mp_hangup
[Feb 10 16:41:07] DEBUG[5901] devicestate.c: Notification of state change to be queued on device/channel MP/1
[Feb 10 16:41:07] DEBUG[5767] devicestate.c: Changing state for MP/1 - state 0 (Unknown)
[Feb 10 16:41:09] VERBOSE[5870] logger.c: -- Remote UNIX connection disconnected
[Feb 10 16:48:28] VERBOSE[6175] logger.c: -- event_dtmf #
[Feb 10 16:48:28] DEBUG[6175] devicestate.c: Notification of state change to be queued on device/channel MP/1
[Feb 10 16:48:28] DEBUG[6173] devicestate.c: Changing state for MP/1 - state 0 (Unknown)
[Feb 10 16:48:29] VERBOSE[6175] logger.c: -- event_dtmf 0
[Feb 10 16:48:29] VERBOSE[6175] logger.c: -- event_dtmf 8
[Feb 10 16:48:29] VERBOSE[6175] logger.c: -- event_dtmf 7
[Feb 10 16:48:29] VERBOSE[6175] logger.c: -- event_dtmf 6
[Feb 10 16:48:30] VERBOSE[6175] logger.c: -- event_dtmf 4
[Feb 10 16:48:30] VERBOSE[6175] logger.c: -- event_dtmf 0
[Feb 10 16:48:31] VERBOSE[6175] logger.c: -- event_dtmf 1
[Feb 10 16:48:31] VERBOSE[6175] logger.c: -- event_dtmf 1
[Feb 10 16:48:31] VERBOSE[6175] logger.c: -- event_dtmf 0
[Feb 10 16:48:31] VERBOSE[6175] logger.c: -- event_dtmf 3
[Feb 10 16:48:34] VERBOSE[6175] logger.c: -- event_digit_timer
[Feb 10 16:48:34] VERBOSE[6175] logger.c: -- extension exists, starting PBX #**********
[Feb 10 16:48:34] DEBUG[6175] devicestate.c: Notification of state change to be queued on device/channel MP/1
[Feb 10 16:48:34] DEBUG[6173] devicestate.c: Changing state for MP/1 - state 0 (Unknown)
[Feb 10 16:48:34] DEBUG[6208] pbx.c: Launching 'Dial'
[Feb 10 16:48:34] VERBOSE[6208] logger.c: -- Executing [#**********@default:1] Dial("MP/1", "SIP/**********@sipaccount|120|r") in new stack
[Feb 10 16:48:34] DEBUG[6208] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw)
[Feb 10 16:48:34] DEBUG[6208] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
[Feb 10 16:48:34] DEBUG[6208] chan_sip.c: Setting NAT on RTP to On
[Feb 10 16:48:34] DEBUG[6208] chan_sip.c: *** Our native formats are 0x2 (gsm)
[Feb 10 16:48:34] DEBUG[6208] chan_sip.c: *** Joint capabilities are 0x0 (nothing)
[Feb 10 16:48:35] DEBUG[6208] chan_sip.c: *** Our capabilities are 0xe (gsm|ulaw|alaw)
[Feb 10 16:48:35] DEBUG[6208] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x2 (gsm)
[Feb 10 16:48:35] DEBUG[6208] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw)
[Feb 10 16:48:35] DEBUG[6208] chan_sip.c: This channel will not be able to handle video.
[Feb 10 16:48:35] DEBUG[6208] rtp.c: Channel 'MP/1' has no RTP, not doing anything
[Feb 10 16:48:35] DEBUG[6208] channel.c: Not copying variable STACK-default-#**********-1.
[Feb 10 16:48:35] DEBUG[6208] chan_sip.c: Outgoing Call for **********
[Feb 10 16:48:35] DEBUG[6208] chan_sip.c: Updating call counter for outgoing call
[Feb 10 16:48:35] DEBUG[6208] chan_sip.c: Our T38 capability (0), joint T38 capability (0)
[Feb 10 16:48:35] DEBUG[6208] chan_sip.c: ** Our capability: 0x6 (gsm|ulaw) Video flag: False
[Feb 10 16:48:35] DEBUG[6208] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
[Feb 10 16:48:35] VERBOSE[6208] logger.c: Audio is at 10.130.1.21 port 17474
[Feb 10 16:48:35] VERBOSE[6208] logger.c: Adding codec 0x4 (ulaw) to SDP
[Feb 10 16:48:35] VERBOSE[6208] logger.c: Adding codec 0x2 (gsm) to SDP
[Feb 10 16:48:35] VERBOSE[6208] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 10 16:48:35] DEBUG[6208] chan_sip.c: -- Done with adding codecs to SDP
[Feb 10 16:48:35] DEBUG[6208] chan_sip.c: Done building SDP. Settling with this capability: 0x6 (gsm|ulaw)
[Feb 10 16:48:35] VERBOSE[6208] logger.c: Reliably Transmitting (NAT) to 66.8.50.218:5060:
INVITE sip:**********@sip.*****.co.za SIP/2.0
Via: SIP/2.0/UDP 10.130.1.21:5060;branch=z9hG4bK26299de5;rport
From: "asterisk" <sip:*********@10.130.1.21>;tag=as279617bc
To: <sip:**********@sip.*****.co.za>
Contact: <sip:*********@10.130.1.21>
Call-ID: 760c6104237ded0448fa3cd2049244e5@10.130.1.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 10 Feb 2011 16:48:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 6208 6208 IN IP4 10.130.1.21
s=session
c=IN IP4 10.130.1.21
t=0 0
m=audio 17474 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Feb 10 16:48:35] DEBUG[6208] sched.c: Attempted to delete nonexistent schedule entry 0!
[Feb 10 16:48:35] VERBOSE[6208] logger.c: -- Called **********@sipaccount
[Feb 10 16:48:35] VERBOSE[6208] logger.c: -- Asked to indicate 'Remote end is ringing' condition on channel MP/1
[Feb 10 16:48:35] VERBOSE[6176] logger.c:
<--- SIP read from 66.8.50.218:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 10.130.1.21:5060;branch=z9hG4bK26299de5;rport=5060
From: "asterisk" <sip:*********@10.130.1.21:5060>;tag=as279617bc
To: <sip:**********@sip.*****.co.za>
Call-ID: 760c6104237ded0448fa3cd2049244e5@10.130.1.21
CSeq: 102 INVITE
Server: OpenSIPS (1.4.2-notls (i386/linux))
Content-Length: 0
<------------->
[Feb 10 16:48:35] VERBOSE[6176] logger.c: --- (8 headers 0 lines) ---
[Feb 10 16:48:35] DEBUG[6176] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '760c6104237ded0448fa3cd2049244e5@10.130.1.21' Request 102: Found
[Feb 10 16:48:35] VERBOSE[6176] logger.c:
<--- SIP read from 66.8.50.218:5060 --->
SIP/2.0 180 Ringing
Record-Route: <sip:66.8.50.218;lr=on;did=b76.de208895>
Via: SIP/2.0/UDP 10.130.1.21:5060;received=10.130.1.21;branch=z9hG4bK26299de5;rport=5060
From: "asterisk" <sip:*********@10.130.1.21:5060>;tag=as279617bc
To: <sip:**********@sip.*****.co.za>;tag=3833144728
Call-ID: 760c6104237ded0448fa3cd2049244e5@10.130.1.21
CSeq: 102 INVITE
Content-Length: 0
<------------->
[Feb 10 16:48:35] VERBOSE[6176] logger.c: --- (8 headers 0 lines) ---
[Feb 10 16:48:35] DEBUG[6176] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '760c6104237ded0448fa3cd2049244e5@10.130.1.21' Request 102: Found
[Feb 10 16:48:35] DEBUG[6176] devicestate.c: Notification of state change to be queued on device/channel SIP/sipaccount-00567448
[Feb 10 16:48:35] VERBOSE[6208] logger.c: -- SIP/sipaccount-00567448 is ringing
[Feb 10 16:48:35] DEBUG[6173] chan_sip.c: Checking device state for peer sipaccount
[Feb 10 16:48:35] DEBUG[6173] devicestate.c: Changing state for SIP/sipaccount - state 1 (Not in use)
[Feb 10 16:48:40] VERBOSE[6176] logger.c:
<--- SIP read from 66.8.50.218:5060 --->
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP 10.130.1.21:5060;received=10.130.1.21;branch=z9hG4bK26299de5;rport=5060
From: "asterisk" <sip:*********@10.130.1.21:5060>;tag=as279617bc
To: <sip:**********@sip.*****.co.za>
Call-ID: 760c6104237ded0448fa3cd2049244e5@10.130.1.21
CSeq: 102 INVITE
Content-Length: 0
<------------->
[Feb 10 16:48:40] VERBOSE[6176] logger.c: --- (7 headers 0 lines) ---
[Feb 10 16:48:40] DEBUG[6176] chan_sip.c: Acked pending invite 102
[Feb 10 16:48:40] DEBUG[6176] chan_sip.c: Stopping retransmission on '760c6104237ded0448fa3cd2049244e5@10.130.1.21' of Request 102: Match Not Found
[Feb 10 16:48:40] VERBOSE[6176] logger.c: -- Got SIP response 486 "Busy here" back from 66.8.50.218
[Feb 10 16:48:40] VERBOSE[6176] logger.c: Transmitting (NAT) to 66.8.50.218:5060:
ACK sip:**********@sip.*****.co.za SIP/2.0
Via: SIP/2.0/UDP 10.130.1.21:5060;branch=z9hG4bK26299de5;rport
From: "asterisk" <sip:*********@10.130.1.21>;tag=as279617bc
To: <sip:**********@sip.*****.co.za>
Contact: <sip:*********@10.130.1.21>
Call-ID: 760c6104237ded0448fa3cd2049244e5@10.130.1.21
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
[Feb 10 16:48:40] DEBUG[6176] chan_sip.c: Setting SIP_ALREADYGONE on dialog 760c6104237ded0448fa3cd2049244e5@10.130.1.21
[Feb 10 16:48:40] VERBOSE[6208] logger.c: -- SIP/sipaccount-00567448 is busy
[Feb 10 16:48:40] DEBUG[6208] channel.c: Hanging up channel 'SIP/sipaccount-00567448'
[Feb 10 16:48:40] DEBUG[6208] chan_sip.c: Hangup call SIP/sipaccount-00567448, SIP callid 760c6104237ded0448fa3cd2049244e5@10.130.1.21)
[Feb 10 16:48:40] DEBUG[6208] devicestate.c: Notification of state change to be queued on device/channel SIP/sipaccount-00567448
[Feb 10 16:48:40] DEBUG[6173] chan_sip.c: Checking device state for peer sipaccount
[Feb 10 16:48:40] DEBUG[6173] devicestate.c: Changing state for SIP/sipaccount - state 1 (Not in use)
[Feb 10 16:48:40] VERBOSE[6208] logger.c: == Everyone is busy/congested at this time (1:1/0/0)
[Feb 10 16:48:40] VERBOSE[6208] logger.c: -- Asked to indicate 'Stop tone' condition on channel MP/1
[Feb 10 16:48:40] DEBUG[6208] rtp.c: Channel 'MP/1' has no RTP, not doing anything
[Feb 10 16:48:40] DEBUG[6208] app_dial.c: Exiting with DIALSTATUS=BUSY.
[Feb 10 16:48:40] VERBOSE[6208] logger.c: == Auto fallthrough, channel 'MP/1' status is 'BUSY'
[Feb 10 16:48:40] VERBOSE[6208] logger.c: -- Asked to indicate 'Remote end is busy' condition on channel MP/1
[Feb 10 16:48:40] DEBUG[6208] devicestate.c: Notification of state change to be queued on device/channel MP/1
[Feb 10 16:48:40] DEBUG[6173] devicestate.c: Changing state for MP/1 - state 0 (Unknown)
[Feb 10 16:48:41] VERBOSE[6176] logger.c: Really destroying SIP dialog '760c6104237ded0448fa3cd2049244e5@10.130.1.21' Method: INVITE
[Feb 10 16:48:42] VERBOSE[6175] logger.c: -- event_onhook
[Feb 10 16:48:42] VERBOSE[6175] logger.c: -- default: hangup sound_on = 1
[Feb 10 16:48:42] DEBUG[6208] channel.c: Soft-Hanging up channel 'MP/1'
[Feb 10 16:48:42] DEBUG[6208] channel.c: Hanging up channel 'MP/1'
[Feb 10 16:48:42] VERBOSE[6208] logger.c: -- start mp_hangup
[Feb 10 16:48:42] DEBUG[6208] devicestate.c: Notification of state change to be queued on device/channel MP/1
[Feb 10 16:48:42] DEBUG[6173] devicestate.c: Changing state for MP/1 - state 0 (Unknown)
[Feb 10 16:48:47] VERBOSE[6178] logger.c: -- Remote UNIX connection disconnected
[Feb 10 16:48:50] DEBUG[6176] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
[Feb 10 16:48:50] VERBOSE[6176] logger.c: Reliably Transmitting (NAT) to 66.8.50.218:5060:
OPTIONS sip:sip.*****.co.za SIP/2.0
Via: SIP/2.0/UDP 10.130.1.21:5060;branch=z9hG4bK49305897;rport
From: "asterisk" <sip:asterisk@10.130.1.21>;tag=as05ff301f
To: <sip:sip.*****.co.za>
Contact: <sip:asterisk@10.130.1.21>
Call-ID: 1b65b678618c02a30a0917e31b429d04@10.130.1.21
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 10 Feb 2011 16:48:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Feb 10 16:48:50] DEBUG[6176] sched.c: Attempted to delete nonexistent schedule entry 0!
[Feb 10 16:48:50] VERBOSE[6176] logger.c:
<--- SIP read from 66.8.50.218:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.130.1.21:5060;branch=z9hG4bK49305897;rport=5060
From: "asterisk" <sip:asterisk@10.130.1.21:5060>;tag=as05ff301f
To: <sip:sip.*****.co.za>;tag=4a858b1e222aafec6644321f187dd12e.b366
Call-ID: 1b65b678618c02a30a0917e31b429d04@10.130.1.21
CSeq: 102 OPTIONS
Server: OpenSIPS (1.4.2-notls (i386/linux))
Content-Length: 0
<------------->
[Feb 10 16:48:50] VERBOSE[6176] logger.c: --- (8 headers 0 lines) ---
[Feb 10 16:48:50] DEBUG[6176] chan_sip.c: Stopping retransmission on '1b65b678618c02a30a0917e31b429d04@10.130.1.21' of Request 102: Match Not Found
[Feb 10 16:48:50] VERBOSE[6176] logger.c: Really destroying SIP dialog '1b65b678618c02a30a0917e31b429d04@10.130.1.21' Method: OPTIONS
--
_____________________________________________________________________
-- 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
