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<DIV><FONT face=Arial size=2><SPAN class=219005904-24062005>I have a new
asterisk install (1.0.7) - and in case it's relevant I'm not using autoload
option in modules.conf. Generally all is working well. However, when
I make a call from my softphone and try to leave a message, the message is
cutoff after a few seconds (whenever I pause for 1 second between words).
</SPAN></FONT><FONT face=Arial size=2><SPAN class=219005904-24062005>Strangely,
when I use an analog phone connected to my ATA, I can record as long as I want
with long pauses. I have VBR and VAD (silence suppression) turned off on
the soft phone.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=219005904-24062005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=219005904-24062005>Here is my SIP debug
output of a call from my softphone to voicemail (ext 232 does not answer).
Can anyone explain the cutoff?</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=219005904-24062005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=219005904-24062005>Thanks</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=219005904-24062005>------------------------------------------------------------------------------------</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>pbx*CLI> sip debug<BR>SIP Debugging
Enabled<BR>pbx*CLI> </FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Sip read: <BR>INVITE sip:232@pbx.ocg.ca
SIP/2.0<BR>To: <sip:232@pbx.ocg.ca><BR>From:
pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660<BR>Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-22694588-1--d87543-;rport<BR>Call-ID:
113d5508a72b5176<BR>CSeq: 1 INVITE<BR>Contact:
<sip:233@172.31.254.106:9330><BR>Max-Forwards: 70<BR>Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<BR>Content-Type:
application/sdp<BR>User-Agent: eyeBeam release 3004t stamp
16741<BR>Content-Length: 270</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=- 7013285 7013368 IN IP4
172.31.254.106<BR>s=eyeBeam<BR>c=IN IP4 172.31.254.106<BR>t=0 0<BR>m=audio 9332
RTP/AVP 100 6 0 8 5 101<BR>a=alt:1 1 : A153D4E1 AFA161AA 172.31.254.106
9332<BR>a=fmtp:101 0-15<BR>a=rtpmap:100 speex/16000<BR>a=rtpmap:101
telephone-event/8000<BR>a=sendrecv</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>12 headers, 11 lines<BR>Using latest request as
basis request<BR>Sending to 172.31.254.106 : 9330 (non-NAT)<BR>Reliably
Transmitting (no NAT):<BR>SIP/2.0 407 Proxy Authentication Required<BR>Via:
SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-22694588-1--d87543-<BR>From:
pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660<BR>To:
<sip:232@pbx.ocg.ca>;tag=as4eb9d1f1<BR>Call-ID: 113d5508a72b5176<BR>CSeq:
1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:232@172.31.254.4><BR>Proxy-Authenticate:
Digest realm="pbx.ocg.ca", nonce="310f6924"<BR>Content-Length: 0</FONT></DIV>
<DIV> </DIV>
<DIV><BR><FONT face=Arial size=2> to 172.31.254.106:9330<BR>Scheduling
destruction of call '113d5508a72b5176' in 15000 ms<BR>Found user
'233'<BR>pbx*CLI> </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sip read: <BR>ACK sip:232@pbx.ocg.ca SIP/2.0<BR>To:
<sip:232@pbx.ocg.ca>;tag=as4eb9d1f1<BR>From:
pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660<BR>Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-22694588-1--d87543-;rport<BR>Call-ID:
113d5508a72b5176<BR>CSeq: 1 ACK<BR>Content-Length: 0</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><BR><FONT face=Arial size=2>7 headers, 0 lines<BR>pbx*CLI> </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sip read: <BR>INVITE sip:232@pbx.ocg.ca
SIP/2.0<BR>To: <sip:232@pbx.ocg.ca><BR>From:
pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660<BR>Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-107041778-1--d87543-;rport<BR>Call-ID:
113d5508a72b5176<BR>CSeq: 2 INVITE<BR>Contact:
<sip:233@172.31.254.106:9330><BR>Max-Forwards: 70<BR>Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<BR>Content-Type:
application/sdp<BR>Proxy-Authorization: Digest
username="233",realm="pbx.ocg.ca",nonce="310f6924",uri="sip:232@pbx.ocg.ca",response="43674ccbcff37fa8066402d8106d0e66",algorithm=MD5<BR>User-Agent:
eyeBeam release 3004t stamp 16741<BR>Content-Length: 270</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=- 7013285 7013368 IN IP4
172.31.254.106<BR>s=eyeBeam<BR>c=IN IP4 172.31.254.106<BR>t=0 0<BR>m=audio 9332
RTP/AVP 100 6 0 8 5 101<BR>a=alt:1 1 : A153D4E1 AFA161AA 172.31.254.106
9332<BR>a=fmtp:101 0-15<BR>a=rtpmap:100 speex/16000<BR>a=rtpmap:101
telephone-event/8000<BR>a=sendrecv</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>13 headers, 11 lines<BR>Using latest request as
basis request<BR>Sending to 172.31.254.106 : 9330 (non-NAT)<BR>Found user
'233'<BR>Found RTP audio format 100<BR>Found RTP audio format 6<BR>Found RTP
audio format 0<BR>Found RTP audio format 8<BR>Found RTP audio format 5<BR>Found
RTP audio format 101<BR>Peer audio RTP is at port 172.31.254.106:9332<BR>Found
description format speex<BR>Found description format
telephone-event<BR>Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x22c
(ulaw|alaw|adpcm|speex)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)<BR>Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723),
combined - 0x1 (g723)<BR>Looking for 232 in menuinternal<BR>list_route: hop:
<sip:233@172.31.254.106:9330><BR>Transmitting (no NAT):<BR>SIP/2.0 100
Trying<BR>Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-107041778-1--d87543-<BR>From:
pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660<BR>To:
<sip:232@pbx.ocg.ca>;tag=as0e1f028d<BR>Call-ID: 113d5508a72b5176<BR>CSeq:
2 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:232@172.31.254.4><BR>Content-Length:
0</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><BR><FONT face=Arial size=2> to
172.31.254.106:9330<BR> -- Executing Macro("SIP/233-a3ba",
"calllocalextension|232|SIP/232|230|Mike Stahl") in new
stack<BR> -- Executing SetVar("SIP/233-a3ba",
"LastStatus=CallDone") in new stack<BR> -- Executing
Playback("SIP/233-a3ba", "/var/lib/asterisk/ocgsounds/pleasewaitwhileitry|skip")
in new stack<BR> -- Executing Dial("SIP/233-a3ba",
"SIP/232|30|r") in new stack<BR>Destroying call </FONT><A
href="mailto:'694cce5213b6205d0df81f4a58d1b670@172.31.254.4'"><FONT face=Arial
size=2>'694cce5213b6205d0df81f4a58d1b670@172.31.254.4'</FONT></A><BR><FONT
face=Arial size=2>Jun 24 00:56:15 NOTICE[7507]: app_dial.c:746 dial_exec: Unable
to create channel of type 'SIP'<BR> == Everyone is busy/congested at this
time<BR> -- Executing NoOp("SIP/233-a3ba", "CHANUNAVAIL") in
new stack<BR> -- Executing Goto("SIP/233-a3ba",
"s-CHANUNAVAIL|1") in new stack<BR> -- Goto
(macro-calllocalextension,s-CHANUNAVAIL,1)<BR> -- Executing
Goto("SIP/233-a3ba", "s-NOANSWER|1") in new stack<BR> -- Goto
(macro-calllocalextension,s-NOANSWER,1)<BR> -- Executing
VoiceMail("SIP/233-a3ba", "u230") in new stack<BR>We're at 172.31.254.4 port
10204<BR>Video is at 172.31.254.4 port 14628<BR>Answering with preferred
capability 0x2 (gsm)<BR>Answering with preferred capability 0x4
(ulaw)<BR>Answering with preferred capability 0x8 (alaw)<BR>Answering with
non-codec capability 0x1 (telephone-event)<BR>Reliably Transmitting (no
NAT):<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-107041778-1--d87543-<BR>From:
pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660<BR>To:
<sip:232@pbx.ocg.ca>;tag=as0e1f028d<BR>Call-ID: 113d5508a72b5176<BR>CSeq:
2 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:232@172.31.254.4><BR>Content-Type:
application/sdp<BR>Content-Length: 261</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=root 7507 7507 IN IP4
172.31.254.4<BR>s=session<BR>c=IN IP4 172.31.254.4<BR>t=0 0<BR>m=audio 10204
RTP/AVP 3 0 8 101<BR>a=rtpmap:3 GSM/8000<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:8
PCMA/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101
0-16<BR>a=silenceSupp:off - - - -</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> to 172.31.254.106:9330<BR>
-- Playing 'vm-theperson' (language 'en')<BR>pbx*CLI> </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sip read: <BR>ACK sip:232@172.31.254.4
SIP/2.0<BR>To: <sip:232@pbx.ocg.ca>;tag=as0e1f028d<BR>From:
pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660<BR>Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-413694733-1--d87543-;rport<BR>Call-ID:
113d5508a72b5176<BR>CSeq: 2 ACK<BR>Contact:
<sip:233@172.31.254.106:9330><BR>Max-Forwards: 70<BR>Proxy-Authorization:
Digest
username="233",realm="pbx.ocg.ca",nonce="310f6924",uri="sip:232@pbx.ocg.ca",response="43674ccbcff37fa8066402d8106d0e66",algorithm=MD5<BR>User-Agent:
eyeBeam release 3004t stamp 16741<BR>Content-Length: 0</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><BR><FONT face=Arial size=2>11 headers, 0 lines<BR>pbx*CLI>
</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sip read: </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><BR><FONT face=Arial size=2>0 headers, 0 lines<BR> --
Playing 'digits/2' (language 'en')<BR> -- Playing 'digits/3'
(language 'en')<BR> -- Playing 'digits/0' (language
'en')<BR> -- Playing 'vm-isunavail' (language
'en')<BR> -- Playing 'vm-intro' (language 'en')<BR>pbx*CLI>
</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sip read: </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><BR><FONT face=Arial size=2>0 headers, 0 lines<BR> --
Playing 'beep' (language 'en')<BR> -- Recording the
message<BR> -- x=0, open writing:
/var/spool/asterisk/voicemail/internalextensions/230/INBOX/msg0008 format:
wav49, 0x8144680<BR>Jun 24 00:56:31 WARNING[7507]: app.c:619
ast_play_and_record: No audio available on SIP/233-a3ba??<BR>
-- User hung up<BR> -- Executing GotoIf("SIP/233-a3ba",
"1?menuinternal|t|2") in new stack<BR> -- Goto
(menuinternal,t,2)<BR> -- Executing Wait("SIP/233-a3ba", "4")
in new stack<BR> -- Executing Goto("SIP/233-a3ba", "s|1") in
new stack<BR> -- Goto (menuinternal,s,1)<BR>
-- Executing GotoIf("SIP/233-a3ba", "0&1?4") in new
stack<BR> -- Executing SetVar("SIP/233-a3ba",
"LastStatus=Try1") in new stack<BR> -- Executing
Goto("SIP/233-a3ba", "11") in new stack<BR> -- Goto
(menuinternal,s,11)<BR> -- Executing
BackGround("SIP/233-a3ba", "/var/lib/asterisk/ocgsounds/enterextension") in new
stack<BR> -- Playing
'/var/lib/asterisk/ocgsounds/enterextension' (language 'en')<BR>pbx*CLI>
</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sip read: </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><BR><FONT face=Arial size=2>0 headers, 0 lines<BR>set_destination: Parsing
<sip:233@172.31.254.106:9330> for address/port to send
to<BR>set_destination: set destination to 172.31.254.106, port 9330<BR>Reliably
Transmitting:<BR>BYE sip:233@172.31.254.106:9330 SIP/2.0<BR>Via: SIP/2.0/UDP
172.31.254.4:5060;branch=z9hG4bK38b4f048;rport<BR>From:
<sip:232@pbx.ocg.ca>;tag=as0e1f028d<BR>To:
pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660<BR>Contact:
<sip:232@172.31.254.4><BR>Call-ID: 113d5508a72b5176<BR>CSeq: 102
BYE<BR>User-Agent: Asterisk PBX<BR>Content-Length: 0</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> (no NAT) to
172.31.254.106:9330<BR>pbx*CLI> </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sip read: <BR>SIP/2.0 200 OK<BR>To:
pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660<BR>From:
<sip:232@pbx.ocg.ca>;tag=as0e1f028d<BR>Via: SIP/2.0/UDP
172.31.254.4:5060;branch=z9hG4bK38b4f048;rport=5060;received=172.31.254.4<BR>Call-ID:
113d5508a72b5176<BR>CSeq: 102 BYE<BR>Contact:
<sip:233@172.31.254.106:9330><BR>Content-Length: 0</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><BR><FONT face=Arial size=2>8 headers, 0 lines<BR>Message is
BYE<BR>Destroying call '113d5508a72b5176'<BR>pbx*CLI> </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sip read: </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><BR><FONT face=Arial size=2>0 headers, 0 lines<BR>pbx*CLI> sip no
debug<BR>SIP Debugging Disabled<BR>pbx*CLI> </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial
size=2>------------------------</FONT></SPAN></DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial size=2>In case it's
relevant, here's my modules.conf. Am I missing something
important?</FONT></SPAN></DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial size=2>[root@pbx asterisk]#
cat modules.conf<BR>; Modules.conf<BR>;</FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial size=2>[modules]
<BR>autoload=no </FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial size=2>; Resources --
<BR>load => res_adsi.so <BR>;load => res_agi.so <BR>;load =>
res_config_odbc.so <BR>load => res_crypto.so <BR>load => res_features.so
<BR>;load => res_indications.so <BR>;load => res_monitor.so <BR>load =>
res_musiconhold.so <BR>;load => res_odbc.so </FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial size=2>; PBX --
<BR>load => pbx_config.so ; Requires N/A <BR>;load => pbx_dundi.so ;
Requires res_crypto.so <BR>;load => pbx_functions.so ; Requires N/A <BR>;load
=> pbx_loopback.so ; Requires N/A <BR>;load => pbx_realtime.so ; Requires
N/A <BR>;load => pbx_spool.so ; Requires N/A </FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial size=2>; Functions --
<BR>;load => func_callerid.so </FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial size=2>; Database
Call Detail Records -- <BR>load => cdr_csv.so ; Requires N/A <BR>;load =>
cdr_custom.so ; Requires N/A <BR>;load => cdr_manager.so ; Requires N/A
<BR>;load => cdr_odbc.so ; Requires N/A <BR>;load => cdr_pgsql.so ;
Requires N/A </FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial size=2>; Channels --
<BR>;load => chan_agent.so ; Requires res_features.so, res_monitor.so,
res_musiconhold.so <BR>;load => chan_features.so ; Requires N/A <BR>load
=> chan_iax2.so ; Requires res_crypto.so, res_features.so <BR>load =>
chan_local.so ; Requires N/A <BR>;load => chan_mgcp.so ; Requires
res_features.so <BR>;load => chan_modem.so ; Requires N/A <BR>;load =>
chan_modem_aopen.so ; Requires chan_modem.so <BR>;load =>
chan_modem_bestdata.so ; Requires chan_modem.so <BR>;load =>
chan_modem_i4l.so ; Requires chan_modem.so <BR>;load => chan_oss.so ;
Requires N/A <BR>;load => chan_phone.so ; Requires N/A <BR>load =>
chan_sip.so ; Requires res_features.so <BR>;load => chan_skinny.so ; Requires
res_features.so </FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial size=2>; Codecs --
<BR>load => codec_a_mu.so ; Requires N/A <BR>load => codec_adpcm.so ;
Requires N/A <BR>load => codec_alaw.so ; Requires N/A <BR>load =>
codec_g726.so ; Requires N/A <BR>load => codec_gsm.so ; Requires N/A <BR>load
=> codec_ilbc.so ; Requires N/A <BR>load => codec_lpc10.so ; Requires N/A
<BR>load => codec_ulaw.so ; Requires N/A </FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial size=2>; Formats --
<BR>;load => format_g723.so ; Requires N/A <BR>load => format_g726.so ;
Requires N/A <BR>;load => format_g729.so ; Requires N/A <BR>load =>
format_gsm.so ; Requires N/A <BR>;load => format_h263.so ; Requires N/A
<BR>load => format_ilbc.so ; Requires N/A <BR>load => format_jpeg.so ;
Requires N/A <BR>load => format_pcm.so ; Requires N/A <BR>load =>
format_pcm_alaw.so ; Requires N/A <BR>;load => format_sln.so ; Requires N/A
<BR>;load => format_vox.so ; Requires N/A <BR>load => format_wav.so ;
Requires N/A <BR>load => format_wav_gsm.so ; Requires N/A
</FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial size=2>; Applications
-- <BR>;load => app_adsiprog.so ; Requires res_adsi.so <BR>;load =>
app_alarmreceiver.so ; Requires N/A <BR>load => app_authenticate.so ;
Requires N/A <BR>load => app_cdr.so ; Requires N/A <BR>load =>
app_chanisavail.so ; Requires N/A <BR>;load => app_chanspy.so ; Requires N/A
<BR>load => app_controlplayback.so ; Requires N/A <BR>;load => app_curl.so
; Requires N/A <BR>;load => app_cut.so ; Requires N/A <BR>;load =>
app_db.so ; Requires N/A <BR>load => app_dial.so ; Requires res_features.so,
res_musiconhold.so <BR>;load => app_dictate.so ; Requires N/A <BR>load =>
app_directory.so ; Requires N/A <BR>;load => app_disa.so ; Requires N/A
<BR>;load => app_dumpchan.so ; Requires N/A <BR>load => app_echo.so ;
Requires N/A <BR>;load => app_enumlookup.so ; Requires N/A <BR>load =>
app_eval.so ; Requires N/A <BR>;load => app_exec.so ; Requires N/A <BR>load
=> app_festival.so ; Requires N/A <BR>load => app_forkcdr.so ; Requires
N/A <BR>;load => app_getcpeid.so ; Requires N/A <BR>;load =>
app_groupcount.so ; Requires N/A <BR>load => app_hasnewvoicemail.so ;
Requires N/A <BR>;load => app_ices.so ; Requires N/A <BR>;load =>
app_image.so ; Requires N/A <BR>;load => app_intercom.so ; Obsolete - does
not load <BR>load => app_lookupblacklist.so ; Requires N/A <BR>load =>
app_lookupcidname.so ; Requires N/A <BR>load => app_macro.so ; Requires N/A
<BR>;load => app_math.so ; Requires N/A <BR>;load => app_md5.so ; Requires
N/A <BR>;load => app_milliwatt.so ; Requires N/A <BR>;load => app_mp3.so ;
Requires N/A <BR>;load => app_nbscat.so ; Requires N/A <BR>;load =>
app_parkandannounce.so ; Requires res_features.so <BR>load => app_playback.so
; Requires N/A <BR>;load => app_privacy.so ; Requires N/A <BR>;load =>
app_queue.so ; Requires res_features.so, res_monitor.so, res_musiconhold.so
<BR>;load => app_random.so ; Requires N/A <BR>;load => app_read.so ;
Requires N/A <BR>;load => app_readfile.so ; Requires N/A <BR>;load =>
app_realtime.so ; Requires N/A <BR>;load => app_record.so ; Requires N/A
<BR>load => app_sayunixtime.so ; Requires N/A <BR>;load => app_senddtmf.so
; Requires N/A <BR>;load => app_sendtext.so ; Requires N/A <BR>load =>
app_setcallerid.so ; Requires N/A <BR>;load => app_setcdruserfield.so ;
Requires N/A <BR>load => app_setcidname.so ; Requires N/A <BR>load =>
app_setcidnum.so ; Requires N/A <BR>;load => app_setrdnis.so ; Requires N/A
<BR>;load => app_settransfercapability.so ; Requires N/A <BR>;load =>
app_sms.so ; Requires N/A <BR>;load => app_softhangup.so ; Requires N/A
<BR>;load => app_striplsd.so ; Requires N/A <BR>;load => app_substring.so
; Requires N/A <BR>;load => app_system.so ; Requires N/A <BR>load =>
app_talkdetect.so ; Requires N/A <BR>;load => app_test.so ; Requires N/A
<BR>;load => app_transfer.so ; Requires N/A <BR>;load => app_txtcidname.so
; Requires N/A <BR>;load => app_url.so ; Requires N/A <BR>;load =>
app_userevent.so ; Requires N/A <BR>load => app_verbose.so ; Requires N/A
<BR>load => app_voicemail.so ; Requires res_adsi.so <BR>;load =>
app_waitforring.so ; Requires N/A <BR>;load => app_waitforsilence.so ;
Requires N/A <BR>;load => app_while.so ; Requires N/A <BR>load =>
app_zapateller.so ; Requires N/A </FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN class=219005904-24062005><FONT face=Arial size=2>[global]
<BR>chan_modem.so=yes </FONT></SPAN></DIV></BODY></HTML>