similar to: username mismatch with 1.6.2.11

Hi. I need to send calls to different programs depending on where the call originates. For example, I need calls from San Diego (NPA 619 and 858) to to be routed differently than L.A. calls. I tried entries like: exten => 4044633/_619.,1,OurApp,sandiego-queue exten => 4044633/_858.,1,OurApp,sandiego-queue exten => 4044633/_213.,1,OurApp,losangeles-queue exten =>

Hmmm.... Can't place calls... Can't access website... Neither of the 3 nameservers answer anything... Anyone heard/know something to explain all this?

*Do FiveThirtyEight, priors and random forests mean more to you than a time of day, entries in a rap sheet, and pristine flora? Read on if that was a yes.* CARD.COM is a well-funded financial startup in Santa Monica, CA. We issue Visa debit cards that connect brands with their fans and enable our customers to make a social statement with their payment. Data and scientific insights are at the core

Here is the situation that I would like to create: Call comes in Receptionist sees that the caller ID is Jenny <8675309> Receptionist picks up phone and transfers call to Batman Batman looks at his phone and sees that the caller ID is Jenny <8675309> I can't seem to figure out how to forward the caller ID. Is this possible with Asterisk?

The Asterisk Development Team has announced the release of Asterisk 1.6.2.11. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.11 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.11. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.11 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

Hi, I have a problem that the machine running asterisk 1.6.2.11 freezes unexpectly time to time. Sometimes it runs for 4 weeks without any problem, sometimes after a free it freezes again in 24 hours. But usually it runs normally for 1 month or so before it freezes again. I could not find any additional info in any file located in /var/log/ including asterisk's messages file. Verbosity

Hello list, I'm using asterisk 1.6.2.11 with realtime SIP (mysql DB). I notice that when the SIP peer registers, the fields 'fullcontact', 'ipaddr', 'port', 'regserver', 'regseconds', 'lastms' are filled with values. But after a while, these fields become empty. Asterisk CLI shows : asterisk*CLI> sip show peers Name/username

Hello list, what is the correct syntax ? exten => s,n,Queue(${queuename},,,,${timeout},cleanpickup.agi^${CHANNEL}) [Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-0000017a': File does not exist. Kind regards, Jonas. -------------- next part -------------- An HTML attachment was

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Hi, I can't figure out what syntax to use with the Dial() "M" parameter for the AEL parser to interpret properly. Creating an AEL macro named "macro-screen()" partly works as a hack, but it must not turn into a gosub properly, so I get warnings about the "return;". Dial(...,tgM(&screen)) with the ael macro named "screen" does not work

Hello list, has anyone else also noticed spontaneous reboots ?! I noticed this today and also yesterday. Can't really see if there is a fixed time between the reboots. Normally al of my SIP peers are registered. When I put up the CLI today I saw that a lot of SIP accounts where UNREACHABLE and needed to register again (what they slowly did). These are realtime SIP peers that reside on

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

I have a voicemail application that users can listen to messages and leave messages. I am looking for a way to play a beep tone to a user when a new message is received when they are on the phone. Here is what I have come up with: in extensions.conf: [beepvoicemail] exten => 1000,1,answer() exten => 1000,2,NoCDR() exten => 1000,3,wait(2) exten => 1000,4,Set(TIMEOUT(absolute)=5)

Hi, The Asterisk Wiki (page: http://www.voip-info.org/wiki/view/Asterisk+func+cdr) mentions I can set any custom CDR field I want. Here is the example it gives: ; Update our accountcode field and then save some random music facts too exten => s,1,Set(CDR(accountcode)=8675309) exten => s,2,Set(CDR(MyFavoriteBand)=Foo Fighters) exten => s,3,Set(CDR(MyFavoriteSong)=Hero) I am

Hi all, I have tried to disable anonymous in smb.conf, by include the global: restrict anonymous = true password server = * encrypt passwords = yes null passwords = no Buy when I ran the : ./smbclient -L myserver -N I have this message: added interface ip=111.xx.xx.xx bcast=111.xx.xx.255 nmask=255.255.252.0 Got a positive name query response from 111.xx.xx.xx( 111.xx.xxx.xx)