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