similar to: MeetMe 'i' option not working correctly?

Guys, this error has been driving me nuts and I see no indication anywhere as to what it may mean. Anybody has any clues on this? -- User ended message by pressing # -- Playing 'auth-thankyou' (language 'en') -- Playing 'vm-review' (language 'en') -- Saving message as is -- Playing 'vm-msgsaved' (language 'en') Mar 4 21:02:06

Hi all, In my Asterisk setup, incoming calls through Cisco PSTN gateway to Asterisk extensions sounds work fine. All calls can be terminated properly after hangup. However, when calls were forwarded to voicemail, after recording & hangup the PSTN calls and cisco FXO port remained connected unless cisco port was manually shut/no shut. # key used to hang up the call did NOT help. Did anyone

Guys. What does this error mean? -- Playing '/var/spool/asterisk/voicemail/intruder/201/unavail' (language 'sp') -- Playing 'vm-intro' (language 'sp') -- Playing 'beep' (language 'sp') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/intruder/201/INBOX/msg0000 format: wav, 0x812b4f0 -- User ended

Dear all, I noticed a very strange problem. When I tried using VoiceMailMain to record my unavailable message, the file does not get created even though I can find the corresponding mssage from asterisk: -- <SIP/2001-b6307d78> Playing 'beep' (language 'en') -- x=0, open writing: /var/spool/asterisk/voicemail/default/2000/unavail.tmp format: wav, 0x82828c8 --

I have a problem with the current trunk code for 1.6.0 as it relates to voicemail. I had the same problem in a previous trunk version as well so I just updated myself to current code - Asterisk SVN-branch-1.6.0-r184281M I have voicemail using ODBC storage. When a new voicemail message is left and the system is, I am guessing, trying to generate the email notification it core dumps. Here is what

i m using asterisk 1.8.3 on a centos 5.5 computer when i try to change my name on the voicemail asterisk dump core here what i got on the console -- User ended message by pressing # -- <SIP/6672-00000002> Playing 'auth-thankyou.alaw' (language 'fr') -- <SIP/6672-00000002> Playing 'vm-review.alaw' (language 'fr') -- Saving message as

I'm having the following problem: I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card) to access voicemail. After dialing the appropriate extension I get voicemail, am presented with the user's unavailable message, and can leave a message normally. The problem comes when I press "#" to end the recording, at which point I am told "Your message has been

Hi, I would like to express my appreciation to your big efforts. I am enjoying Asterisk very much. My Asterisk works very well, but I encountered a segfault at voicemail after pressing # to end the recording. Please see the log below. My asterisk is running on RedHat 8. When booting the Asterisk, I found a WARNING around IAX, and it says "Unable to open IAX timing interface: No such

Hello list, I wonder if someone might be able to clear up something for me. I recently set up asterisk and have now managed to get the MeetMe application up and running. When I dial the extension to access the conference/MeetMe application, the only prompt I hear is:"You are currently the only person in this conference." When I use a friend's newly installed asterisk, I hear:

Skipped content of type multipart/alternative-------------- next part -------------- ############ # amd BOX # ############ ## Step 1 ## Bob(ext. 6202) place a remote IAX2 call to the operator (ext. 6302) ## Reminder : _62XX are register on 'amd' and _63XX on 'dell' -- Executing SetGroup("SIP/6202-d193", "IAX") in new stack -- Executing

The Asterisk Development Team has announced the release of Asterisk 11.18.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.18.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.18.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.18.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hello, I am running Samba 3.0.28 on a Solaris 10 8/07 x86_64 machine. Directory services and authentication are provided by MS Active Directory's LDAP and Kerberos. My problem is that, when attempting to add an ACE to a file from Windows (XP, SP2), I get an error saying "Access Denied". The share is a directory on a ZFS filesystem (NFSv4 ACLs). Below is the share definition:

I recently setup app_meetme with the 'i' option. My boss wants users to say their name and go directly into the conference instead of reviewing the recording. If anyone else is interested in this behavior becoming an option, has a suggestion what letter to use as the option (I was thinking 'i' -- with review and 'I' -- without review), or anything else, I'd appreciate

The Asterisk Development Team has announced the release of Asterisk 13.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.4.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

The Asterisk Development Team has announced the release of Asterisk 13.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.4.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

On Tue, Jan 07, 2014 at 11:16:08AM +0100, Pino Toscano wrote: > On Friday 27 December 2013 10:58:15 you wrote: > > virt-sysprep either runs with all default operations or a selected > > list of operations with the --enable argument. A few times I've > > found I'd like to use the default list, but minus one or two > > operations in particular, however there's

--Scraw_of_Flies_285_000 Content-Type: TEXT/plain; charset=us-ascii Content-MD5: eXeT31BJngKeovsqhTpOHg== Dear R-project, I am having difficulty compiling R for my unix machine. Attached is the config.log file that has all the necessary info. Can you help? Thank you. Mutlu.. ------------------------------- Mr. Mutlu Ozdogan Center for Remote Sensing Boston University 725 Commonwealth Avenue

On Friday 27 December 2013 10:58:15 you wrote: > virt-sysprep either runs with all default operations or a selected > list of operations with the --enable argument. A few times I've > found I'd like to use the default list, but minus one or two > operations in particular, however there's no easy way to specify > this. > > A --disable argument that took the

While I know this is not a true asterisk problem, I figure someone where may know. When you click on Messages and it gives you the count of Urgent, New, etc. How can you make the phone gather that information? For example, my phone shows me there is an e-mail. It also sends an e-mail. Yet, when I click on message before I connect to the contact center, it doesn't have any counts. Here is