Dave
2006-Mar-20 15:29 UTC
[Asterisk-Users] Asterisk Disconnecting after 30sec when someone leaving VM
Hello, I have started having a strange problem.
Asterisk is connected via 4 analog lines to PSTN and
we have SIP phones internally. All was working fine
but recently each time a user calls from PSTN and when
he is leaving a voicemail for someone, the caller gets
disconnected after 30 secs. We have AMP installed.
This is reproducible and is happening always. It seems
that Asterisk is disconnecting because of a "silence"
of 10 secs. But that is not case because the caller is
still leaving a message. When calling from a SIP to
SIP phone, this issue does not occur.
Here is a log:
-----
-- Executing VoiceMail("Zap/3-1", "u533") in new
stack
-- Playing
'/var/spool/asterisk/voicemail/default/533/temp'
(language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/533/INBOX/msg00
-- x=1, open writing:
/var/spool/asterisk/voicemail/default/533/INBOX/msg00
-- x=2, open writing:
/var/spool/asterisk/voicemail/default/533/INBOX/msg00
-- Recording automatically stopped after a silence
of 10 seconds
-- Playing 'auth-thankyou' (language 'en')
-- Executing Hangup("Zap/3-1", "") in new stack
== Spawn extension (macro-vm, s-NOANSWER, 2) exited
non-zero on 'Zap/3-1' in m
== Spawn extension (macro-exten-vm, s, 8) exited
non-zero on 'Zap/3-1' in macr
== Spawn extension (aa_1, 240, 1) exited non-zero on
'Zap/3-1'
-- Executing Hangup("Zap/3-1", "") in new stack
== Spawn extension (aa_1, h, 1) exited non-zero on
'Zap/3-1'
-- Hungup 'Zap/3-1'
-- Hungup 'Zap/1-1'
---------------
Any pointers/suggestions will be appreciated.
Dave
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William M Conlon
2006-Mar-20 15:56 UTC
[Asterisk-Users] Asterisk Disconnecting after 30sec when someone leaving VM
I had very mysterious hangups of PSTN outbound calls which I traced to zapata.conf. I had increased rx_gain = 10 and tx_gain = 10 because volume was always low because Polycom phone was always resetting the volume. Anyway, I patched my sip.cfg for Polycom so my volume would stay at the user's setting, and reset the zap gains to 0, and my hangups went away. [Maybe the high gain was causing an output to saturate and a small amount of silence looked like a polarity reversal - SWAG]. On Mar 20, 2006, at 2:29 PM, Dave wrote:> Hello, I have started having a strange problem. > Asterisk is connected via 4 analog lines to PSTN and > we have SIP phones internally. All was working fine > but recently each time a user calls from PSTN and when > he is leaving a voicemail for someone, the caller gets > disconnected after 30 secs. We have AMP installed. > This is reproducible and is happening always. It seems > that Asterisk is disconnecting because of a "silence" > of 10 secs. But that is not case because the caller is > still leaving a message. When calling from a SIP to > SIP phone, this issue does not occur. > > > > Here is a log: > ----- > > -- Executing VoiceMail("Zap/3-1", "u533") in new > stack > -- Playing > '/var/spool/asterisk/voicemail/default/533/temp' > (language 'en') > -- Playing 'vm-intro' (language 'en') > -- Playing 'beep' (language 'en') > -- Recording the message > -- x=0, open writing: > /var/spool/asterisk/voicemail/default/533/INBOX/msg00 > -- x=1, open writing: > /var/spool/asterisk/voicemail/default/533/INBOX/msg00 > -- x=2, open writing: > /var/spool/asterisk/voicemail/default/533/INBOX/msg00 > -- Recording automatically stopped after a silence > of 10 seconds > -- Playing 'auth-thankyou' (language 'en') > -- Executing Hangup("Zap/3-1", "") in new stack > > == Spawn extension (macro-vm, s-NOANSWER, 2) exited > non-zero on 'Zap/3-1' in m > == Spawn extension (macro-exten-vm, s, 8) exited > non-zero on 'Zap/3-1' in macr > == Spawn extension (aa_1, 240, 1) exited non-zero on > 'Zap/3-1' > -- Executing Hangup("Zap/3-1", "") in new stack > == Spawn extension (aa_1, h, 1) exited non-zero on > 'Zap/3-1' > -- Hungup 'Zap/3-1' > -- Hungup 'Zap/1-1' > > --------------- > > Any pointers/suggestions will be appreciated. > > Dave > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersBill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:bill@tothept.com web: http://www.tothept.com