Michael Stahl
2005-Jun-23 22:05 UTC
[Asterisk-Users] Voicemail recording cutoff when silent for 1 second
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). 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.
Here is my SIP debug output of a call from my softphone to voicemail
(ext 232 does not answer). Can anyone explain the cutoff?
Thanks
------------------------------------------------------------------------
------------
pbx*CLI> sip debug
SIP Debugging Enabled
pbx*CLI>
Sip read:
INVITE sip:232@pbx.ocg.ca SIP/2.0
To: <sip:232@pbx.ocg.ca>
From: pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660
Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-22694588-1--d87543-;rport
Call-ID: 113d5508a72b5176
CSeq: 1 INVITE
Contact: <sip:233@172.31.254.106:9330>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 3004t stamp 16741
Content-Length: 270
v=0
o=- 7013285 7013368 IN IP4 172.31.254.106
s=eyeBeam
c=IN IP4 172.31.254.106
t=0 0
m=audio 9332 RTP/AVP 100 6 0 8 5 101
a=alt:1 1 : A153D4E1 AFA161AA 172.31.254.106 9332
a=fmtp:101 0-15
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
12 headers, 11 lines
Using latest request as basis request
Sending to 172.31.254.106 : 9330 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-22694588-1--d87543-
From: pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660
To: <sip:232@pbx.ocg.ca>;tag=as4eb9d1f1
Call-ID: 113d5508a72b5176
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:232@172.31.254.4>
Proxy-Authenticate: Digest realm="pbx.ocg.ca",
nonce="310f6924"
Content-Length: 0
to 172.31.254.106:9330
Scheduling destruction of call '113d5508a72b5176' in 15000 ms
Found user '233'
pbx*CLI>
Sip read:
ACK sip:232@pbx.ocg.ca SIP/2.0
To: <sip:232@pbx.ocg.ca>;tag=as4eb9d1f1
From: pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660
Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-22694588-1--d87543-;rport
Call-ID: 113d5508a72b5176
CSeq: 1 ACK
Content-Length: 0
7 headers, 0 lines
pbx*CLI>
Sip read:
INVITE sip:232@pbx.ocg.ca SIP/2.0
To: <sip:232@pbx.ocg.ca>
From: pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660
Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-107041778-1--d87543-;rport
Call-ID: 113d5508a72b5176
CSeq: 2 INVITE
Contact: <sip:233@172.31.254.106:9330>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest
username="233",realm="pbx.ocg.ca",nonce="310f6924",uri="sip:232@pbx.ocg.
ca",response="43674ccbcff37fa8066402d8106d0e66",algorithm=MD5
User-Agent: eyeBeam release 3004t stamp 16741
Content-Length: 270
v=0
o=- 7013285 7013368 IN IP4 172.31.254.106
s=eyeBeam
c=IN IP4 172.31.254.106
t=0 0
m=audio 9332 RTP/AVP 100 6 0 8 5 101
a=alt:1 1 : A153D4E1 AFA161AA 172.31.254.106 9332
a=fmtp:101 0-15
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
13 headers, 11 lines
Using latest request as basis request
Sending to 172.31.254.106 : 9330 (non-NAT)
Found user '233'
Found RTP audio format 100
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 5
Found RTP audio format 101
Peer audio RTP is at port 172.31.254.106:9332
Found description format speex
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x22c
(ulaw|alaw|adpcm|speex)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for 232 in menuinternal
list_route: hop: <sip:233@172.31.254.106:9330>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-107041778-1--d87543-
From: pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660
To: <sip:232@pbx.ocg.ca>;tag=as0e1f028d
Call-ID: 113d5508a72b5176
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:232@172.31.254.4>
Content-Length: 0
to 172.31.254.106:9330
-- Executing Macro("SIP/233-a3ba",
"calllocalextension|232|SIP/232|230|Mike Stahl") in new stack
-- Executing SetVar("SIP/233-a3ba",
"LastStatus=CallDone") in new
stack
-- Executing Playback("SIP/233-a3ba",
"/var/lib/asterisk/ocgsounds/pleasewaitwhileitry|skip") in new stack
-- Executing Dial("SIP/233-a3ba", "SIP/232|30|r") in new
stack
Destroying call '694cce5213b6205d0df81f4a58d1b670@172.31.254.4'
<mailto:'694cce5213b6205d0df81f4a58d1b670@172.31.254.4'>
Jun 24 00:56:15 NOTICE[7507]: app_dial.c:746 dial_exec: Unable to create
channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing NoOp("SIP/233-a3ba", "CHANUNAVAIL") in new
stack
-- Executing Goto("SIP/233-a3ba", "s-CHANUNAVAIL|1") in
new stack
-- Goto (macro-calllocalextension,s-CHANUNAVAIL,1)
-- Executing Goto("SIP/233-a3ba", "s-NOANSWER|1") in new
stack
-- Goto (macro-calllocalextension,s-NOANSWER,1)
-- Executing VoiceMail("SIP/233-a3ba", "u230") in new
stack
We're at 172.31.254.4 port 10204
Video is at 172.31.254.4 port 14628
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-107041778-1--d87543-
From: pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660
To: <sip:232@pbx.ocg.ca>;tag=as0e1f028d
Call-ID: 113d5508a72b5176
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:232@172.31.254.4>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 7507 7507 IN IP4 172.31.254.4
s=session
c=IN IP4 172.31.254.4
t=0 0
m=audio 10204 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 172.31.254.106:9330
-- Playing 'vm-theperson' (language 'en')
pbx*CLI>
Sip read:
ACK sip:232@172.31.254.4 SIP/2.0
To: <sip:232@pbx.ocg.ca>;tag=as0e1f028d
From: pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660
Via: SIP/2.0/UDP
172.31.254.106:9330;branch=z9hG4bK-d87543-413694733-1--d87543-;rport
Call-ID: 113d5508a72b5176
CSeq: 2 ACK
Contact: <sip:233@172.31.254.106:9330>
Max-Forwards: 70
Proxy-Authorization: Digest
username="233",realm="pbx.ocg.ca",nonce="310f6924",uri="sip:232@pbx.ocg.
ca",response="43674ccbcff37fa8066402d8106d0e66",algorithm=MD5
User-Agent: eyeBeam release 3004t stamp 16741
Content-Length: 0
11 headers, 0 lines
pbx*CLI>
Sip read:
0 headers, 0 lines
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/3' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
pbx*CLI>
Sip read:
0 headers, 0 lines
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/internalextensions/230/INBOX/msg0008
format: wav49, 0x8144680
Jun 24 00:56:31 WARNING[7507]: app.c:619 ast_play_and_record: No audio
available on SIP/233-a3ba??
-- User hung up
-- Executing GotoIf("SIP/233-a3ba",
"1?menuinternal|t|2") in new
stack
-- Goto (menuinternal,t,2)
-- Executing Wait("SIP/233-a3ba", "4") in new stack
-- Executing Goto("SIP/233-a3ba", "s|1") in new stack
-- Goto (menuinternal,s,1)
-- Executing GotoIf("SIP/233-a3ba", "0&1?4") in new
stack
-- Executing SetVar("SIP/233-a3ba", "LastStatus=Try1")
in new stack
-- Executing Goto("SIP/233-a3ba", "11") in new stack
-- Goto (menuinternal,s,11)
-- Executing BackGround("SIP/233-a3ba",
"/var/lib/asterisk/ocgsounds/enterextension") in new stack
-- Playing '/var/lib/asterisk/ocgsounds/enterextension' (language
'en')
pbx*CLI>
Sip read:
0 headers, 0 lines
set_destination: Parsing <sip:233@172.31.254.106:9330> for address/port
to send to
set_destination: set destination to 172.31.254.106, port 9330
Reliably Transmitting:
BYE sip:233@172.31.254.106:9330 SIP/2.0
Via: SIP/2.0/UDP 172.31.254.4:5060;branch=z9hG4bK38b4f048;rport
From: <sip:232@pbx.ocg.ca>;tag=as0e1f028d
To: pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660
Contact: <sip:232@172.31.254.4>
Call-ID: 113d5508a72b5176
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 172.31.254.106:9330
pbx*CLI>
Sip read:
SIP/2.0 200 OK
To: pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660
From: <sip:232@pbx.ocg.ca>;tag=as0e1f028d
Via: SIP/2.0/UDP
172.31.254.4:5060;branch=z9hG4bK38b4f048;rport=5060;received=172.31.254.
4
Call-ID: 113d5508a72b5176
CSeq: 102 BYE
Contact: <sip:233@172.31.254.106:9330>
Content-Length: 0
8 headers, 0 lines
Message is BYE
Destroying call '113d5508a72b5176'
pbx*CLI>
Sip read:
0 headers, 0 lines
pbx*CLI> sip no debug
SIP Debugging Disabled
pbx*CLI>
------------------------
In case it's relevant, here's my modules.conf. Am I missing something
important?
[root@pbx asterisk]# cat modules.conf
; Modules.conf
;
[modules]
autoload=no
; Resources --
load => res_adsi.so
;load => res_agi.so
;load => res_config_odbc.so
load => res_crypto.so
load => res_features.so
;load => res_indications.so
;load => res_monitor.so
load => res_musiconhold.so
;load => res_odbc.so
; PBX --
load => pbx_config.so ; Requires N/A
;load => pbx_dundi.so ; Requires res_crypto.so
;load => pbx_functions.so ; Requires N/A
;load => pbx_loopback.so ; Requires N/A
;load => pbx_realtime.so ; Requires N/A
;load => pbx_spool.so ; Requires N/A
; Functions --
;load => func_callerid.so
; Database Call Detail Records --
load => cdr_csv.so ; Requires N/A
;load => cdr_custom.so ; Requires N/A
;load => cdr_manager.so ; Requires N/A
;load => cdr_odbc.so ; Requires N/A
;load => cdr_pgsql.so ; Requires N/A
; Channels --
;load => chan_agent.so ; Requires res_features.so, res_monitor.so,
res_musiconhold.so
;load => chan_features.so ; Requires N/A
load => chan_iax2.so ; Requires res_crypto.so, res_features.so
load => chan_local.so ; Requires N/A
;load => chan_mgcp.so ; Requires res_features.so
;load => chan_modem.so ; Requires N/A
;load => chan_modem_aopen.so ; Requires chan_modem.so
;load => chan_modem_bestdata.so ; Requires chan_modem.so
;load => chan_modem_i4l.so ; Requires chan_modem.so
;load => chan_oss.so ; Requires N/A
;load => chan_phone.so ; Requires N/A
load => chan_sip.so ; Requires res_features.so
;load => chan_skinny.so ; Requires res_features.so
; Codecs --
load => codec_a_mu.so ; Requires N/A
load => codec_adpcm.so ; Requires N/A
load => codec_alaw.so ; Requires N/A
load => codec_g726.so ; Requires N/A
load => codec_gsm.so ; Requires N/A
load => codec_ilbc.so ; Requires N/A
load => codec_lpc10.so ; Requires N/A
load => codec_ulaw.so ; Requires N/A
; Formats --
;load => format_g723.so ; Requires N/A
load => format_g726.so ; Requires N/A
;load => format_g729.so ; Requires N/A
load => format_gsm.so ; Requires N/A
;load => format_h263.so ; Requires N/A
load => format_ilbc.so ; Requires N/A
load => format_jpeg.so ; Requires N/A
load => format_pcm.so ; Requires N/A
load => format_pcm_alaw.so ; Requires N/A
;load => format_sln.so ; Requires N/A
;load => format_vox.so ; Requires N/A
load => format_wav.so ; Requires N/A
load => format_wav_gsm.so ; Requires N/A
; Applications --
;load => app_adsiprog.so ; Requires res_adsi.so
;load => app_alarmreceiver.so ; Requires N/A
load => app_authenticate.so ; Requires N/A
load => app_cdr.so ; Requires N/A
load => app_chanisavail.so ; Requires N/A
;load => app_chanspy.so ; Requires N/A
load => app_controlplayback.so ; Requires N/A
;load => app_curl.so ; Requires N/A
;load => app_cut.so ; Requires N/A
;load => app_db.so ; Requires N/A
load => app_dial.so ; Requires res_features.so, res_musiconhold.so
;load => app_dictate.so ; Requires N/A
load => app_directory.so ; Requires N/A
;load => app_disa.so ; Requires N/A
;load => app_dumpchan.so ; Requires N/A
load => app_echo.so ; Requires N/A
;load => app_enumlookup.so ; Requires N/A
load => app_eval.so ; Requires N/A
;load => app_exec.so ; Requires N/A
load => app_festival.so ; Requires N/A
load => app_forkcdr.so ; Requires N/A
;load => app_getcpeid.so ; Requires N/A
;load => app_groupcount.so ; Requires N/A
load => app_hasnewvoicemail.so ; Requires N/A
;load => app_ices.so ; Requires N/A
;load => app_image.so ; Requires N/A
;load => app_intercom.so ; Obsolete - does not load
load => app_lookupblacklist.so ; Requires N/A
load => app_lookupcidname.so ; Requires N/A
load => app_macro.so ; Requires N/A
;load => app_math.so ; Requires N/A
;load => app_md5.so ; Requires N/A
;load => app_milliwatt.so ; Requires N/A
;load => app_mp3.so ; Requires N/A
;load => app_nbscat.so ; Requires N/A
;load => app_parkandannounce.so ; Requires res_features.so
load => app_playback.so ; Requires N/A
;load => app_privacy.so ; Requires N/A
;load => app_queue.so ; Requires res_features.so, res_monitor.so,
res_musiconhold.so
;load => app_random.so ; Requires N/A
;load => app_read.so ; Requires N/A
;load => app_readfile.so ; Requires N/A
;load => app_realtime.so ; Requires N/A
;load => app_record.so ; Requires N/A
load => app_sayunixtime.so ; Requires N/A
;load => app_senddtmf.so ; Requires N/A
;load => app_sendtext.so ; Requires N/A
load => app_setcallerid.so ; Requires N/A
;load => app_setcdruserfield.so ; Requires N/A
load => app_setcidname.so ; Requires N/A
load => app_setcidnum.so ; Requires N/A
;load => app_setrdnis.so ; Requires N/A
;load => app_settransfercapability.so ; Requires N/A
;load => app_sms.so ; Requires N/A
;load => app_softhangup.so ; Requires N/A
;load => app_striplsd.so ; Requires N/A
;load => app_substring.so ; Requires N/A
;load => app_system.so ; Requires N/A
load => app_talkdetect.so ; Requires N/A
;load => app_test.so ; Requires N/A
;load => app_transfer.so ; Requires N/A
;load => app_txtcidname.so ; Requires N/A
;load => app_url.so ; Requires N/A
;load => app_userevent.so ; Requires N/A
load => app_verbose.so ; Requires N/A
load => app_voicemail.so ; Requires res_adsi.so
;load => app_waitforring.so ; Requires N/A
;load => app_waitforsilence.so ; Requires N/A
;load => app_while.so ; Requires N/A
load => app_zapateller.so ; Requires N/A
[global]
chan_modem.so=yes
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050624/6524d13f/attachment.htm
Reasonably Related Threads
Wisdom of the Ancients
