similar to: Re: Asterisk-Users digest, Vol 1 #286 - 14 msgs

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Thanks Ben, Adam and Petr for the feedback! So currently things that need to be done for the Monitor resource are: 1. Name files uniquely. Adam, your naming suggestion is great. I think we should stick with that, with a minor change: I don't think we should put destination channel name in the file names. In some instances there will be no destination channels (plain IVR: play, record, dtmf),

Mahmut, excellent summary :-). I look forward to your next update. One little thing, In the manager events that show start/stop monitoring, can you please include a field that indicates the filename(s) to which the monitoring was written? Thanks, Ben -----Original Message----- From: Fettahlioglu, Mahmut [mailto:Mahmut.Fettahlioglu@oa.com.au] Sent: Tuesday, April 15, 2003 5:17 AM To:

Hi all, I'm using asterisk in a production environment now and this afternoon I got reports complaining that it was not working. Looking at the asterisk console output, I saw it contains lots of error messages as printed below. Unfortunately it is not obvious from the logs as to what started all this. Just before the error messages start, everything seems to be working fine with no problems.

Mahmut, First of all, I'd like to reiterate what a great patch this has been. I'd also like to voice support for having an option to mix the files on the fly, and name them uniquely. While I was able to smoothly put the files together with soxmix, I see on the fly mixing as hugely beneficial to an automated solution to saving/delivering the messages without intervention. One feature,

----- Original Message ----- > Subject: Re: [Asterisk-Users] Analog Phones with Status Light Indicators > From: Adam Goryachev <mailinglists@websitemanagers.com.au> > To: asterisk-users@lists.digium.com > Organization: Website Managers > Date: Thu, 12 Aug 2004 14:53:02 +1000 > Reply-To: asterisk-users@lists.digium.com > > On Wed, 2004-08-11 at 20:42, Steven

Hi Wade, Sorry for replying so late. I had been sucked into other tasks for a while and only now can catch up with the list. > When I dial my iaxtel number from my extension on channel > Zap/15, I get two > files recorded in /var/spool/asterisk/monitor: > > Zap-15-1-in.wav and Zap-15-1-out.wav and they sound fine. > > When I dial again, it overwrites the same two files.

On Wed, Jun 6, 2012 at 5:33 PM, Alexey Samsonov <samsonov at google.com> wrote: > Hi, Chatsiri! > > >> ---------- Forwarded message ---------- >> From: Chatsiri Ratana <insiderboy at gmail.com> >> Date: Wed, Jun 6, 2012 at 2:15 PM >> Subject: [LLVMdev] Compile-rt throw error undeclared identifier >> 'O_CLOEXEC' >> To: llvmdev at

Hi, A question about transfert. How can I make transfert with the the person who call. X call Z and X transfert Z to Y. I only succeed to do X call Z and Z transfert to Y. If someone have a solution it will be very good =) regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I notice that i don't have callerID in my Voimail when someone drop me a message from H323 Client. Is there a tip to have this CallerID ? Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030804/af4203ca/attachment.htm

Hi, I use with asterisk gnugk a gatekeeper for h323 client. I don't understand why asterisk can't have the H323-ID (callerID). In the gatekeeper's monitor I have this H323-ID but not in asterisk. Does anyone know something about it, or how can I send a caller ID to asterisk ? Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I try to use ISDN4Linux drivers with Asterisk. In modem.conf i put /dev/ttyIO. Everything is OK when i lauch asterisk but, when i call Asterisk nothing happen. Someone can help me ? Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030407/13e78d8e/attachment.htm

hi everybody, Is there SIP client which work with Asterisk and can be embedded in a HTML page ? Thanks Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031029/226a8b1b/attachment.htm

hi, I use IAX softphone with asterisk and I notice that a call between two IAX softphones end after 1 min. Then I can't hear anything but the call still in progress. I have this log in asterisk IAX debug: Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 21589 DCall: 00001 [192.168.1.22:4569] Tx-Frame Retry[000] --

hi, how can I add or remove this line "include => context" by the command CLI ? regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030827/979ddd76/attachment.htm

The details for the Australian cisco ATA186 are below: > -----Original Message----- > From: Tony Du [mailto:tony.du@action.com.au] > Sent: Tuesday, 8 July 2003 4:31 PM > To: 'Adam Goryachev' > Subject: RE: [Asterisk-Users] ATA 186 in Australia > > Hi Adam, > > I sold a Cisco ATA186 I1 2 port adaptor (Cisco code: > SW-SMH-UL-ATA-2P)to you on 16/10/02) >

Hi, I try to use conferences with Asterisk but i'm not succeed in doing this. I have set in meetme.conf : conf => 4000 and add an extension in extension.conf. But i have Asterisk WARNING Unable to open pseudo channel. Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, Someone know how to install conference (MeetMe) without a zaptel devices ? Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030414/f8a9140a/attachment.htm

Hi, I have a little question, I use asterisk with Netmeeting client. When I call netmeeting client with a phone. I don\'t have his ID in netmeeting window i have something like : ???;..dhz instead of 28. Someone know a way to display this ID ? Thanks you so much Rattana