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I have had a search for this, but didn't come up with any results, so maybe I am using the wrong terms, sorry if this is an FAQ. For those who want to forward their incoming voice calls to a mobile, it could be a cheaper option to call a mobile from another mobile on the same network. This probably wouldn't be useful for users in USA, Canada or Hong Kong as costs to call a mobile is

Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't have any scope rules I have to find another way to do what I want. Thanks in advance.

Hello, I recently updated my asterisk-addons-1.6.2 to the last revision and I have this problem that I don't know how to interpret, bug or not. I connected a Nokia N80 phone to use chan_mobile and everything works great until the phone starts getting disconnected after the call finished and sometimes during the call attempt. Is this a bug or a possible known issue for Nokia phones? # rpm

Hello, I am looking for a dialer program, free or not, that allows me to perform scheduled calls, generate reports and let me upload sound files. Is there something that fits these features?. If there is not any product like I mentioned before I am interested to build this kind of software but I need ideas to make it useful for technical and non-technical people. I don't want to spend my

Hello list, I want to know the options that I have as an Asterisk user to make a call and control its progress. I have been using the spool directory to originate my calls using this format: "Channel: " "Callerid: " "MaxRetries:" "RetryTime: " "WaitTime: " "Context: " "Extension: " "Priority: " Is there any other

Hello, When I attempt to make a call using AMI interface with originate action I successfully specify all of the needed parameters but when I try to control the flow of the call I am unable to identify each call because asterisk uses some kind of unique identification appended to the channel string. E.g. channel: SIP/1000 results in SIP/1000-*0845ea38*. I also found an auto-generated unique

Hello list, I wonder what are the ideas to improve chan_mobile implementation to make it usb compliant. I am highly interested as a developer and I want to know if there is any initial project. I read part of the chan_mobile source code and a TS of hands-profile-protocol trying to understand better the environment. After that, I started some research about the possibility of doing the same

Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is

Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090802/abb21767/attachment.htm