rendered paste body# more /etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
span=2,1,0,ccs,hdb3
bchan=32-46,48-62
dchan=47
loadzone = us
defaultzone = us
===============================================================================
# more /etc/asterisk/zapata.conf
[channels]
switchtype=euroisdn
echocancel=yes
echocancelwhenbridged=no
callgroup=1
pickupgroup=1
echotrainig=no
busydetect=no
callprogress=no
pridialplan=unknown
localpridialplan=national
immediate=no
overlapdial=yes
signalling=pri_cpe
group=1
context=default
channel => 1-15,17-31
signalling = pri_cpe
group=2
context=default
channel => 32-46,48-62
===============================================================================
DEBUG LOG:
=================> Initial Conection to IVR
Jun 6 12:28:49 VERBOSE[19571] logger.c: -- Accepting call from '82623436' to '0500' on channel 0/7, span 1
Jun 6 12:28:49 DEBUG[19571] chan_zap.c: Enabled echo cancellation on channel 7
Jun 6 12:28:49 VERBOSE[22707] logger.c: -- Executing NoOp("Zap/7-1", "Zap/g2") in new stack
Jun 6 12:28:49 VERBOSE[22707] logger.c: -- Executing Set("Zap/7-1", "LIN=30") in new stack
Jun 6 12:28:49 DEBUG[22707] pbx.c: Function result is '0'
.....
=================> Conection to SIP/Extention
Jun 6 12:28:55 VERBOSE[22707] logger.c: -- Executing Dial("Zap/7-1", "SIP/1172|18|wWtT|") in new stack
Jun 6 12:28:55 DEBUG[22707] chan_sip.c: Setting NAT on RTP to 0
Jun 6 12:28:55 DEBUG[22707] chan_sip.c: Outgoing Call for 1172
Jun 6 12:28:55 DEBUG[22707] chan_sip.c: Call to user '1172' is 1 out of 1
Jun 6 12:28:55 VERBOSE[22707] logger.c: We're at 192.168.20.239 port 12494
Jun 6 12:28:55 VERBOSE[22707] logger.c: Adding codec 0x4 (ulaw) to SDP
Jun 6 12:28:55 VERBOSE[22707] logger.c: Adding codec 0x400 (ilbc) to SDP
Jun 6 12:28:55 VERBOSE[22707] logger.c: Adding codec 0x2 (gsm) to SDP
Jun 6 12:28:55 VERBOSE[22707] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Jun 6 12:28:55 VERBOSE[22707] logger.c: 13 headers, 12 lines
Jun 6 12:28:55 VERBOSE[22707] logger.c: Reliably Transmitting (no NAT) to 192.168.20.155:10882:
INVITE sip:1172@192.168.20.155:10882;rinstance=4ad92b42430b0796 SIP/2.0^M
Via: SIP/2.0/UDP 192.168.20.239:5060;branch=z9hG4bK08ac1ff6;rport^M
From: "82623436" <sip:82623436@192.168.20.239>;tag=as0bf10d0b^M
To: <sip:1172@192.168.20.155:10882;rinstance=4ad92b42430b0796>^M
Contact: <sip:82623436@192.168.20.239>^M
Call-ID: 69bfd202182232e919c477f30eac791e@192.168.20.239^M
CSeq: 102 INVITE^M
User-Agent: Asterisk^M
Max-Forwards: 70^M
Date: Tue, 06 Jun 2006 17:28:55 GMT^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
Content-Type: application/sdp^M
Content-Length: 269^M
^M
v=0^M
o=root 19554 19554 IN IP4 192.168.20.239^M
s=session^M
c=IN IP4 192.168.20.239^M
t=0 0^M
m=audio 12494 RTP/AVP 0 97 3 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:97 iLBC/8000^M
a=rtpmap:3 GSM/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=silenceSupp:off - - - -^M
---
Jun 6 12:28:55 VERBOSE[22707] logger.c: -- Called 1172
Jun 6 12:28:55 DEBUG[22422] chan_sip.c: Setting NAT on RTP to 0
Jun 6 12:28:55 DEBUG[22422] chan_sip.c: Outgoing Call for 1159
Jun 6 12:28:55 VERBOSE[22422] logger.c: -- Called SIP/1159
Jun 6 12:28:55 VERBOSE[19574] logger.c:
=================> CALL IS ENDED:
Jun 6 12:29:49 VERBOSE[19571] logger.c: -- Channel 0/7, span 1 got hangup request
Jun 6 12:29:49 DEBUG[22707] channel.c: Got a FRAME_CONTROL (8) frame on channel Zap/7-1
Jun 6 12:29:49 DEBUG[22707] channel.c: Bridge stops bridging channels Zap/7-1 and SIP/1172-42ab
Jun 6 12:29:49 DEBUG[22707] chan_sip.c: update_call_counter(1172) - decrement call limit counter
Jun 6 12:29:49 DEBUG[22707] chan_sip.c: Call to user '1172' removed from call limit 1
Jun 6 12:29:49 VERBOSE[22707] logger.c: set_destination: Parsing <sip:1172@192.168.20.155:10882;rinstance=4ad92b42430b0796> for address/port to send to
Jun 6 12:29:49 VERBOSE[22707] logger.c: set_destination: set destination to 192.168.20.155, port 10882
Jun 6 12:29:49 VERBOSE[22707] logger.c: Reliably Transmitting (no NAT) to 192.168.20.155:10882:
BYE sip:1172@192.168.20.155:10882;rinstance=4ad92b42430b0796 SIP/2.0^M
Via: SIP/2.0/UDP 192.168.20.239:5060;branch=z9hG4bK46ea62d7;rport^M
From: "82623436" <sip:82623436@192.168.20.239>;tag=as0bf10d0b^M
To: <sip:1172@192.168.20.155:10882;rinstance=4ad92b42430b0796>;tag=d106166a^M
Contact: <sip:82623436@192.168.20.239>^M
Call-ID: 69bfd202182232e919c477f30eac791e@192.168.20.239^M
CSeq: 103 BYE^M
User-Agent: Asterisk^M
Max-Forwards: 70^M
Content-Length: 0^M
^M
---
Jun 6 12:29:49 DEBUG[22707] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Jun 6 12:29:49 VERBOSE[22707] logger.c: == Spawn extension (macro-superdial, s, 21) exited non-zero on 'Zap/7-1' in macro 'superdial'
Jun 6 12:29:49 VERBOSE[22707] logger.c: == Spawn extension (macro-superdial, s, 21) exited non-zero on 'Zap/7-1'
Jun 6 12:29:49 DEBUG[22707] pbx.c: Function result is '82623436'
Jun 6 12:29:49 DEBUG[22707] pbx.c: Function result is '82623436'