rendered paste body[Jul 23 03:44:11] NOTICE[5744] manager.c: Unable to open management configuration manager.conf. Call management disabled.
[Jul 23 03:44:11] NOTICE[5744] cdr.c: CDR simple logging enabled.
[Jul 23 03:44:11] NOTICE[5744] loader.c: 135 modules will be loaded.
[Jul 23 03:44:11] WARNING[5744] res_musiconhold.c: Cannot open dir /usr/local/asterisk/share/asterisk/sounds/moh/custom/delme2 or dir does not exist
[Jul 23 03:44:11] NOTICE[5744] res_smdi.c: Unable to load config smdi.conf: SMDI disabled
[Jul 23 03:44:11] WARNING[5744] res_smdi.c: No SMDI interfaces are available to listen on, not starting SDMI listener.
[Jul 23 03:44:12] NOTICE[5744] chan_agent.c: No agent configuration found -- agent support disabled
[Jul 23 03:44:12] WARNING[5744] config.c: Unknown directive 'qualify=yes' at line 19 of /usr/local/asterisk/etc/asterisk/iax.conf
[Jul 23 03:44:12] NOTICE[5744] iax2-provision.c: No IAX provisioning configuration found, IAX provisioning disabled.
[Jul 23 03:44:12] NOTICE[5744] chan_mgcp.c: Unable to load config mgcp.conf, MGCP disabled
[Jul 23 03:44:12] NOTICE[5744] chan_oss.c: Unable to load config oss.conf
[Jul 23 03:44:12] WARNING[5744] config.c: Unknown directive 'qualify=yes' at line 48 of /usr/local/asterisk/etc/asterisk/sip.conf
[Jul 23 03:44:12] WARNING[5744] config.c: Unknown directive '' at line 57 of /usr/local/asterisk/etc/asterisk/sip.conf
[Jul 23 03:44:12] WARNING[5744] config.c: Unknown directive 'qualify=yes' at line 68 of /usr/local/asterisk/etc/asterisk/sip.conf
[Jul 23 03:44:12] WARNING[5744] config.c: Unknown directive 'qualify=yes' at line 77 of /usr/local/asterisk/etc/asterisk/sip.conf
[Jul 23 03:44:12] WARNING[5744] config.c: Unknown directive 'qualify=yes' at line 87 of /usr/local/asterisk/etc/asterisk/sip.conf
[Jul 23 03:44:12] WARNING[5744] config.c: Unknown directive 'qualify=yes' at line 97 of /usr/local/asterisk/etc/asterisk/sip.conf
[Jul 23 03:44:12] WARNING[5744] config.c: Unknown directive 'qualify=yes' at line 107 of /usr/local/asterisk/etc/asterisk/sip.conf
[Jul 23 03:44:12] WARNING[5744] config.c: Unknown directive 'qualify=yes' at line 116 of /usr/local/asterisk/etc/asterisk/sip.conf
[Jul 23 03:44:12] NOTICE[5744] chan_skinny.c: Unable to load config skinny.conf, Skinny disabled
[Jul 23 03:44:12] NOTICE[5744] pbx_ael.c: Starting AEL load process.
[Jul 23 03:44:12] NOTICE[5744] pbx_ael.c: AEL load process: calculated config file name '/usr/local/asterisk/etc/asterisk/extensions.ael'.
[Jul 23 03:44:12] NOTICE[5744] pbx_ael.c: File /usr/local/asterisk/etc/asterisk/extensions.ael not found; AEL declining load
[Jul 23 03:44:12] WARNING[5744] config.c: Unknown directive 'exten' at line 25 of /usr/local/asterisk/etc/asterisk/extensions.conf
[Jul 23 03:44:12] ERROR[5744] pbx_dundi.c: Unable to load config dundi.conf
[Jul 23 03:44:12] ERROR[5744] app_amd.c: Configuration file amd.conf missing.
[Jul 23 03:44:12] WARNING[5744] app_festival.c: No such configuration file festival.conf
[Jul 23 03:44:12] WARNING[5744] app_followme.c: No follow me config file (followme.conf), so no follow me
[Jul 23 03:44:12] NOTICE[5744] app_queue.c: No call queueing config file (queues.conf), so no call queues
[Jul 23 03:44:12] WARNING[5744] translate.c: plc_samples 160 format 6
[Jul 23 03:44:12] WARNING[5744] translate.c: plc_samples 160 format 6
[Jul 23 03:44:12] WARNING[5744] translate.c: plc_samples 160 format 6
[Jul 23 03:44:12] WARNING[5744] translate.c: plc_samples 160 format 6
[Jul 23 03:44:12] WARNING[5744] translate.c: plc_samples 160 format 6
[Jul 23 03:44:12] WARNING[5744] translate.c: plc_samples 180 format 6
[Jul 23 03:44:12] WARNING[5744] translate.c: plc_samples 160 format 6
[Jul 23 03:44:12] WARNING[5744] cdr_csv.c: unable to load config: cdr.conf
[Jul 23 03:44:12] WARNING[5744] cdr_custom.c: Failed to load configuration file. Module not activated.
[Jul 23 03:44:22] DEBUG[5744] chan_iax2.c: We don't do requested format gsm, falling back to peer capability 1542
[Jul 23 03:44:22] VERBOSE[5744] logger.c: -- Accepting AUTHENTICATED call from 10.1.0.52:
> requested format = gsm,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (),
> priority = mine
[Jul 23 03:44:22] VERBOSE[5744] logger.c: -- Executing [0505391157@family:1] Goto("IAX2/mahmoud-2", "outbound|0505391157|1") in new stack
[Jul 23 03:44:22] VERBOSE[5744] logger.c: -- Goto (outbound,0505391157,1)
[Jul 23 03:44:22] VERBOSE[5744] logger.c: -- Executing [0505391157@outbound:1] Log("IAX2/mahmoud-2", "NOTICE|### "" <> Trying to Call ###") in new stack
[Jul 23 03:44:22] NOTICE[5744] Ext. 0505391157: ### "" <> Trying to Call ###
[Jul 23 03:44:22] VERBOSE[5744] logger.c: -- Executing [0505391157@outbound:2] Answer("IAX2/mahmoud-2", "") in new stack
[Jul 23 03:44:22] VERBOSE[5744] logger.c: -- Executing [0505391157@outbound:3] Authenticate("IAX2/mahmoud-2", "1111") in new stack
[Jul 23 03:44:22] VERBOSE[5744] logger.c: -- <IAX2/mahmoud-2> Playing 'agent-pass' (language 'en')
[Jul 23 03:44:23] VERBOSE[5744] logger.c: Really destroying SIP dialog '1e9dd3a90ada4a5a024e8bb87bfdb734@127.0.0.1' Method: NOTIFY
[Jul 23 03:44:23] DEBUG[5744] chan_iax2.c: Ooh, voice format changed to 4
[Jul 23 03:44:26] VERBOSE[5744] logger.c: -- <IAX2/mahmoud-2> Playing 'auth-thankyou' (language 'en')
[Jul 23 03:44:27] VERBOSE[5744] logger.c: -- Executing [0505391157@outbound:4] Log("IAX2/mahmoud-2", "NOTICE|### "" <> Passed Auth and Calling ###") in new stack
[Jul 23 03:44:27] NOTICE[5744] Ext. 0505391157: ### "" <> Passed Auth and Calling ###
[Jul 23 03:44:27] VERBOSE[5744] logger.c: -- Executing [0505391157@outbound:5] Dial("IAX2/mahmoud-2", "SIP/pots/0505391157") in new stack
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Video is at 10.1.0.1 port 29402
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Audio is at 10.1.0.1 port 15550
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x4 (ulaw) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x8 (alaw) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x2 (gsm) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x10 (g726aal2) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x20 (adpcm) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x40 (slin) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x80 (lpc10) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x400 (ilbc) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x800 (g726) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x10000 (jpeg) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x20000 (png) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x40000 (h261) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x80000 (h263) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x100000 (h263p) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding codec 0x200000 (h264) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Reliably Transmitting (NAT) to 10.1.0.3:5060:
INVITE sip:0505391157@10.1.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.1:5060;branch=z9hG4bK4020ea1b;rport
From: "Unknown" <sip:Unknown@10.1.0.1>;tag=as744223e6
To: <sip:0505391157@10.1.0.3>
Contact: <sip:Unknown@10.1.0.1>
Call-ID: 0a45b52d6dffa39b4cab570c666bc472@10.1.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 23 Jul 2007 03:44:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 648
v=0
o=root 5744 5744 IN IP4 10.1.0.1
s=session
c=IN IP4 10.1.0.1
b=CT:384
t=0 0
m=audio 15550 RTP/AVP 0 8 3 112 5 10 7 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=se[Jul 23 03:44:27] VERBOSE[5744] logger.c: -- Called pots/0505391157
[Jul 23 03:44:27] VERBOSE[5744] logger.c:
<--- SIP read from 10.1.0.3:5060 --->
SIP/2.0 404 Not Found
To: <sip:0505391157@10.1.0.3>;tag=9ff16fd78472e90i0
From: "Unknown" <sip:Unknown@10.1.0.1>;tag=as744223e6
Call-ID: 0a45b52d6dffa39b4cab570c666bc472@10.1.0.1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.1.0.1:5060;branch=z9hG4bK4020ea1b
Server: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0
<------------->
[Jul 23 03:44:27] VERBOSE[5744] logger.c: --- (8 headers 0 lines) ---
[Jul 23 03:44:27] VERBOSE[5744] logger.c: Transmitting (NAT) to 10.1.0.3:5060:
ACK sip:0505391157@10.1.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.1:5060;branch=z9hG4bK4020ea1b;rport
From: "Unknown" <sip:Unknown@10.1.0.1>;tag=as744223e6
To: <sip:0505391157@10.1.0.3>;tag=9ff16fd78472e90i0
Contact: <sip:Unknown@10.1.0.1>
Call-ID: 0a45b52d6dffa39b4cab570c666bc472@10.1.0.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
[Jul 23 03:44:27] VERBOSE[5744] logger.c: -- SIP/pots-086ea000 is circuit-busy
[Jul 23 03:44:27] VERBOSE[5744] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
[Jul 23 03:44:27] VERBOSE[5744] logger.c: == Auto fallthrough, channel 'IAX2/mahmoud-2' status is 'CONGESTION'
[Jul 23 03:44:28] VERBOSE[5744] logger.c: Really destroying SIP dialog '0a45b52d6dffa39b4cab570c666bc472@10.1.0.1' Method: INVITE
[Jul 23 03:44:37] DEBUG[5744] chan_iax2.c: We're hanging up IAX2/mahmoud-2 now...
[Jul 23 03:44:37] VERBOSE[5744] logger.c: -- Hungup 'IAX2/mahmoud-2'