Regarding this message, I've actually been told one caller who has
consistently had this problem was using Vonage, but calling from his
Verizon line, it worked. This skewed my survey.
Therefore I do believe it's the same callers having the issue, and in
which case, I think Vonage is to blame.
I found this thread:
http://forums.digium.com/viewtopic.php?p=49236&highlight=&sid=d3888f3bb90e5c96b5c0432bd632a2d4
but it doesn't help much.
All incoming calls are using IAX.
Did anyone have a similar problem and resolve it?
Thank you.
Leah Newmark
Capalon VoIP
asterisk-users-request at lists.digium.com wrote:
>Message: 8
>Date: Thu, 19 Jul 2007 10:41:44 -0400
>From: Leah Newmark <lnewmark at capalon.com>
>Subject: [asterisk-users] Blank Voicemails
>To: asterisk-users at lists.digium.com
>Message-ID: <469F7828.8000403 at capalon.com>
>Content-Type: text/plain; charset=ISO-8859-1
>
>Hi, we're running Asterisk 1.2.10 and have been randomly being left
>blank voicemails with long messages that we can't hear.
>
>I've searched and searched but cannot find a solution.
>
>This is what happens:
>Internal Server runs Asterisk 1.2.10 where our mailboxes are
>Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are
>bridged between this server and our internal server.
>
>I have not heard any complaints from users on the .13 server, but it's
>happening too frequently to call a fluke on the .10 server.
>Caller gets voicemail, leaves a message, hangs up. Voicemail message is
>emailed to user saying the correct length (0:32, 1:12, etc.), tries to
>play it and player says 0 seconds long. Tries to access via phone, and
>the message again is blank, even though the text file specifies correct
>length.
>Voicemail is being saved in .WAV (wav49).
>
>I tried adding in
>[options]
>transmit_silence_during_record = yes
>into asterisk.conf and it seemed to help for a bit, but then we started
>getting the odd behavior again.
>
>Here is a capture of a failed message:
>//DIDN'T WORK
>Jul 6 11:57:07 DEBUG[9601] app.c: Locked path
>'/var/spool/asterisk/voicemail/default/116/INBOX'
>Jul 6 11:57:07 DEBUG[9601] app.c: Unlocked path
>'/var/spool/asterisk/voicemail/default/116/INBOX'
>Jul 6 11:57:07 DEBUG[9601] app.c: play_and_record: <None>,
>/var/spool/asterisk/voicemail/default/116/tmp/IVnRHt, 'wav49'
>Jul 6 11:57:55 DEBUG[9601] app.c: Locked path
>'/var/spool/asterisk/voicemail/default/116/INBOX'
>Jul 6 11:57:55 DEBUG[9601] app.c: Unlocked path
>'/var/spool/asterisk/voicemail/default/116/INBOX'
>Jul 6 11:57:55 DEBUG[9601] app_voicemail.c: Attaching file
>'/var/spool/asterisk/voicemail/default/116/INBOX/msg0000', format
'WAV',
>uservm is '2048', global is 2048
>Jul 6 11:57:55 DEBUG[9601] app_voicemail.c: Sent mail to
>username at mydomain.com with command '/usr/sbin/sendmail -t'
>
>//THIS WORKED/WORKED
>Jul 6 12:11:24 DEBUG[10184] app.c: Locked path
>'/var/spool/asterisk/voicemail/default/116/INBOX'
>Jul 6 12:11:24 DEBUG[10184] app.c: Unlocked path
>'/var/spool/asterisk/voicemail/default/116/INBOX'
>Jul 6 12:11:24 DEBUG[10184] app.c: play_and_record: <None>,
>/var/spool/asterisk/voicemail/default/116/tmp/rGc1XJ, 'wav49'
>Jul 6 12:11:51 DEBUG[10184] app.c: Locked path
>'/var/spool/asterisk/voicemail/default/116/INBOX'
>Jul 6 12:11:51 DEBUG[10184] app.c: Unlocked path
>'/var/spool/asterisk/voicemail/default/116/INBOX'
>Jul 6 12:11:51 DEBUG[10184] app_voicemail.c: Attaching file
>'/var/spool/asterisk/voicemail/default/116/INBOX/msg0000', format
'WAV',
>uservm is '2048', global is 2048
>Jul 6 12:11:51 DEBUG[10184] app_voicemail.c: Sent mail to
>username at mydomain.com with command '/usr/sbin/sendmail -t'
>
>They look identical! Same mailbox, same debug output, different behavior.
>
>I was noticing a pattern of certain callers (which made me turn on the
>record silence option), but my users tell me it's not only those
>callers, and sometimes those callers do successfully leave messages; I
>only hear when it doesn't work.
>
>What can I do?! I'm stumped, and the situation is intolerable.
>
>Thanks!
>
>Leah Newmark
>Capalon VoIP
>
>
>
>
>------------------------------
>
>ade Procedure (Yusuf)
> 3. Re: New book "Asterisk Cookbook" any good? (Andrew Latham)
> 4. Re: how to use call transfer (satish patel)
> 5. Re: Sip Providers (Anthony Francis)
> 6. Re: how to use call transfer (satish patel)
> 7. Re: open up firewall ports for Asterisk - safe? (David Gomillion)
> 8. Blank Voicemails (Leah Newmark)
> 9. Re: Pass Dialed number to a script (Jared Smith)
> 10. Re: Parsing IAXPeers from Asterisk Manager (PHP API) (Jared Smith)
> 11. Re: G729 copy protection (Jared Smith)
> 12. Re: Upgrade Procedure (Nitesh Divecha)
> 13. Re: Gtalk/Jabber connect issues in 1.4.8 (Bruce Ferrell)
> 14. Re: Upgrade Procedure (Jared Smith)
> 15. Re: 1.4.X howto disable able xpp with ./configure (Tzafrir Cohen)
> 16. Re: G729 copy protection (Bruce McAlister)
> 17. Re: G729 copy protection (Bruce McAlister)
> 18. PRI Card (mail-lists)
> 19. Re: G729 copy protection (Jared Smith)
> 20. Re: how to use call transfer (Gordon Henderson)
> 21. Re: New book "Asterisk Cookbook" any good? (Kristian
Kielhofner)
> 22. Re: PRI Card (Jared Smith)
> 23. Re: Sip Providers (Al Bochter)
> 24. Re: how to use call transfer (Bruno De Luca)
>
>
>----------------------------------------------------------------------
>
>Message: 1
>Date: Thu, 19 Jul 2007 16:11:15 +0200
>From: <jan.sarin at securia.se>
>Subject: Re: [asterisk-users] Zap channels unavailable?
>To: <asterisk-users at lists.digium.com>
>Message-ID:
> <0FF4F1903968F943B5EA2521CD5296C16EEF36 at exchange.securia.local>
>Content-Type: text/plain; charset="us-ascii"
>
>Hi,
>
>I was talking to a technican at our telco yesterday and he told me that
>this problem was most likely caused by our PBX sending "channel
>identification" Exclusive when we dial out. If there's a heavy load
and
>someone is dialing in on the same time on the same channel that we try
>to dial out from - it causes a deadlock. He said some Cisco PBXs have
>the same problem.
>
>Now, I'm no asterisk expert and I don't quite understand what this
>means. I've emailed the list asking if this can be changed to Preferred
>or Negotiation as the technican told me to. But I got no response yet.
>
>I did however "solve" the problem by reversing the channels that
we dial
>out from (so now it tries the last channel first and then backwards to
>the first). Since all of our incoming calls come from the first to the
>last this minimizes the risk of a "collision" of the
incoming/outgoing
>calls. This is of cource no long-term solution but anyway.
>
>I need to know if it's possible to change "channel
identification"
>(whatever that is) to preferred or negotiation.
>
>Regards,
>Jan
>
>
>
>Martin Smith wrote:
>
>Hello Jan,
>
>We have also been seeing this issue, and we are running Asterisk
>1.2.17/Zaptel 1.2.16/LibPRI 1.2.4-r2. We have been informed by our PRI
>provider that a "3rd party vendor" has applied firmware to some
hardware
>along our path, and that it has an unfortunate bug of hanging B-channels
>in the PRI flags "resetting" state. We have been assured that the
vendor
>has been given a 30-day deadline (starting maybe 2 weeks ago) to fix the
>problem, and that it will go away soon. In the mean time, we've also had
>to restart Asterisk to free our B-channels for use, and any link-level
>event potentially re-hangs them.
>
>Keep us posted if you find out anything!
>
>Martin Smith, Systems Developer
>martins at bebr.ufl.edu
>Bureau of Economic and Business Research
>University of Florida
>(352) 392-0171 Ext. 221
>
>
>
>
>
>>-----Original Message-----
>>From: asterisk-users-bounces at lists.digium.com
>>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
>>jan.sarin at securia.se
>>Sent: Tuesday, July 17, 2007 9:44 AM
>>To: asterisk-users at lists.digium.com
>>Subject: [asterisk-users] Zap channels unavailable?
>>
>>Hi,
>>
>>Lately we've noticed that some Zap channels on one of our PRIs are
>>unavailable. We have 2 PRI lines with 60 channels in total.
>>On the first
>>PRI there are currently 20 channels that are not being used for some
>>reason.
>>
>>I tried googling around and found some similar problems but
>>there really
>>was no solution (?). I'm not sure if this problem has occured now
>>because of more load on the pbx but the machine should take
>>it just fine
>>(2x3,0 ghz xeon with 1 gb ram etc).
>>
>>Restarting asterisk makes the zaps' available again but they get
>>"locked" later again. It seems it's always the same
channels that are
>>unavailable too?
>>
>>This one is unavailable and not being used... It's been in PRI Flags
>>state "resetting" for hours now.
>>
>>Channel: 1
>>File Descriptor: 11
>>Span: 1
>>Extension:
>>Dialing: no
>>Context: from-pstn
>>Caller ID: 702821667
>>Calling TON: 33
>>Caller ID name:
>>Destroy: 0
>>InAlarm: 0
>>Signalling Type: PRI Signalling
>>Radio: 0
>>Owner: <None>
>>Real: <None>
>>Callwait: <None>
>>Threeway: <None>
>>Confno: -1
>>Propagated Conference: -1
>>Real in conference: 0
>>DSP: no
>>Relax DTMF: no
>>Dialing/CallwaitCAS: 0/0
>>Default law: alaw
>>Fax Handled: no
>>Pulse phone: no
>>Echo Cancellation: 128 taps unless TDM bridged, currently OFF
>>PRI Flags: Resetting
>>PRI Logical Span: Implicit
>>Hookstate (FXS only): Onhook
>>
>>If anyone can help me with this I'd be really glad. Thanks.
>>
>>Regards,
>>Jan
>>
>>_______________________________________________
>>--Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>>asterisk-users mailing list
>>To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>
>
>
>------------------------------
>
>Message: 2
>Date: Thu, 19 Jul 2007 16:14:46 +0200
>From: Yusuf <yusuf at ecntelecoms.com>
>Subject: Re: [asterisk-users] Upgrade Procedure
>To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
>Message-ID: <469F71D6.2080302 at ecntelecoms.com>
>Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more
information
>X-ECN Telecoms-MailScanner: Found to be clean
>X-ECN Telecoms-MailScanner-From: yusuf at ecntelecoms.com
>X-Spam-Status: No
>
>Nitesh Divecha wrote:
>
>
>>Hello All,
>>
>>I would like to upgrade my recently installed Asterisk 1.2.21.1 to
>>Asterisk 1.4.8?
>>
>>My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6
>>05:25:07 EDT 2007 i686 i686 i386 GNU/Linux
>>
>>Is there any detail step by step procedure to uninstall the current
>>version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons
>>1.4.2?
>>
>>Cheers,
>>Nitesh
>>
>>
>
>
>Hi,
>
>there is an UPGRADE.txt file in each folder of asterisk, zaptel, etc.
>You now need to './configure' before 'make'. Also check out
'make menuselect' to select
>which modules you need or don't. Please check out the default configs
first, look in
>asterisk-1.4.8/configs/
>
>
>
>
>
