search for: journo

Hi, I was wondering if anyone is successfully running Asterisk in a cloud environment. If you could state which cloud you are using, I'd appreciate it. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/076ff188/attachment.htm

Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050920/1ce45adb/attachment.htm

...IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150 per month, please contact Kesher Communications. ________________________________ Dan Journo IT Business Consultant Kesher Communications Ltd Tel: 07957 233 599 Web: http://www.KesherCommunications.com <http://www.keshercommunications.com/> Live Chat/Instant Support: Click Here <http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7F508C6B61A91700343&lang=en...

...p?id=14426 I am using the "s" option with ChanIsAvail because if I run the page, it interrupts the current call. I need to give the caller the busy signal when a minimum of 1 channel is in use. Can anyone suggest how to test whether the user is on the phone? Many thanks Dan Journo Kesher Communications Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091103/d93e6123/attachment.htm

Hi, Does anyone have any advice on customising firmware of an SPA921 so that it can be locked to a sip provider and display logos on the config pages. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f6aa2510/attachment.htm

Is there a guy somewhere on how much bandwidth each codec uses, along with the advantages and disadvantages of each one? Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050923/3bee2776/attachment.htm

...ver. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091012/9a94904b/attachment.htm

Does anyone know how to rewind the music on hold? Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060202/ab26b18f/attachment.htm

Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -------------- next part

...IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150 per month, please contact Kesher Communications. ________________________________ Dan Journo IT Business Consultant Kesher Communications Ltd Tel: 07957 233 599 Web: http://www.KesherCommunications.com <http://www.keshercommunications.com/> Live Chat/Instant Support: Click Here <http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7F508C6B61A91700343&lang=en...

Hi, We have SPA921 handsets which apparently support Paging, however i can't find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/bef6b4f0/attachment.htm

...n still speak to each other without having to wait for one of the 2 other calls to end. I thought that maybe one way would be to duplicate the outbound sip settings and label them "outbound_client_1" and then use call-limit within that. Has anyone got any experience of this? Thanks Dan Journo Kesher Communications Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091206/7f39cef7/attachment.htm

Hi, Im looking at using MixMonitor to record calls and I know that I need to set the filename first. However, with the number of calls coming in, hard coding the filename isnt an option. So I need to do something like this:- MixMonitor(RANDOMNUMBER.wav) But can't find a way to generate a random number. I thought that maybe I could use a unique variable that already exists for the current

Hi, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? Thanks Dan --------------

Hi, Which DTMF mode do people mostly use? I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user. So if I call a company that has a menu system, I can't use the menu. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I think ive found a bug but need someone to double check. Whenever I issue a "reload" in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a similar problem? Asterisk 1.4.32 Mysql realtime. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed...

Hi, I think I already know the answer to this question, but is there any way to do the following using realtime? Or do I have to create a full dialplan for each client without using includes? [client1_phones] include => client1_internal include => client1_outgoing_calls include => test_calls include => parkedcalls [client2_phones] include => client2_internal include =>

Hi, I'm trying to view a list of the active calls to see if I can restart Asterisk. When I do 'sip show channels', I get a huge list like this (just a sample pasted):- 92.110.7.210 (None) 198827f2469 00102/00000 0x0 (nothing) No Init: OPTIONS 92.110.7.210 (None) 6b211bb04ac 00102/00000 0x0 (nothing) No Init: OPTIONS 92.108.34.153

Hi, Does anyone know how to play music to a caller while you dial a second call? Once the second calls has answered, i'd like to music to stop, and the calls to be bridged. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060130/eb3402df/attachment.htm

...lowing at point 3?:- Lookup the realtime users db and read the MailBox column for that buddy. If the mailbox column is empty, play a message saying "Sorry, no one is available." If the column has data in it, do the following:- exten => _11XXXX,3,VoiceMail(MailBoxID) Many thanks Dan Journo http://www.TextOver.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060404/6fc3f8db/attachment.htm