similar to: Diaplan reload command not working

Hi, Is the dev_state can also be used to track a mobile phone's status via SIP? I tried it on several phones(nokia, samsung) but it returns NOANSWER but i can hear a beep beep beep sound indicating that it is currently busy. regards, RYAN ICASIANO

Hi all, I'm trying to get outgoing fax working from an SPA2102 to a PSTN fax machine via a T.38 enabled trunk.? I've got t38pt_udptl = yes faxdetect=no in my sip.conf file.? The ATA has all of the T.38 options turned on, echo cancellation is off, as well as silence suppression off.? The only configured codec is u711.? When the user tries to send a fax, it gets to the point where it

I have an Asterisk with multiple IP's, on the same subnet. When a call comes in, I need to send it back out via SIP, but need that only one IP is used as originating IP for all calls. For example machines has 192.168.50.3 192.168.50.4 192.168.50.5 .... but when I originate the second leg of a call, the IP address that is supposed to be read as source IP must be 192.168.50.5, regardless of how

For the 3rd consecutive term, the US Senate has introduced the "Truth in caller ID Act of 2009". It was passed by the Senate (finally) in January, and has moved to the House for a vote. A lot of states have ambiguous or overly restrictive language on how caller ID may be manipulated. For instance, if you have a PBX, and a call comes in from the PSTN, which you then loop back out

Hi, I'm trying to send Voice mails from my existing Windows application to an Asterisk system. I'm new to Asterisk and to VOIP. Could you please guide me with this? Regards, Shyamala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101108/3238c1d5/attachment.htm

Hi gang, In 500 words or less (if possible), please explain what is a legal music-on-hold file? My boss hates the stuff provided with the distribution and I figure that I'm asking for trouble if I take my Les Mis tracks and run them through Audacity and SOX to make new files. Thanks in advance Danny Nicholas -------------- next part -------------- An HTML attachment was

Hello, For an (old) Asterisk 1.4, how can I tell from the dialplan that a parking lot has available slots, or that a parking slot is empty ? Shall I just park the call with Park() application and , for instance, program next priority as if it would be triggered when the parking lot is full ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Is there a way to remove a global variable from dialplan ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090615/03d38624/attachment.htm

Hi all, When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately. At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc. Thanks in advance, Giang -------------- next part -------------- An HTML

We intend to use Asterisk with a very large dialplan (with a lot of functionality for 3000+ users). Each user will be able to change several of his parameters in the dialplan, so we will be forced to reload the diaplan constantly. Has anybody else any previous experience with a similar installation? There are some things that we'd like to know, if anybody can help us. These are: - Is

Is there an easy way to setup diaplan so when someone pushes a digit such as * during a call, they will be transferred to another destination. For example, a caller is hearing ringing while calling a UA, but instead of waiting for the UA to pick up, they can push * and go directly to that UA's voicemail. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I`m occassionally seeing CDR(accountcode)'s value empty at a place in my diaplan where it was filled with some value a few lines before, with nothing else having changed it. It`s giving me headaches (as I rely on it for MySQL queries). Anything I can do? Mick -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Does asterisk func odbc support multi query? I'm executing stored procedure which returns two tables. With tsql command I can see both tables. But asterisk only shows the first. My database is MSSQL. Maybe there is workaround... Thanks -- Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I think this have been talked over several times but I couldn't find any answer. Sorry for asking. I want from dialplan, to transfer a callee to a context-extension-priority that would play a given fax file to callee (callee is supposed to be a fax number). I can get caller's channel id (with built-in CHANNEL variable). I found BRIDGEPEER but its value remains unset (see bellow)

Hi, I have a little situation with my dialplan, and I am wondering if what I want is even possible. Here it is: I have three contexts, context1 includes contexts2, and context2 includes context3. In other words, in context1 all extensions of context2 and context3 are valid (and actually working, so that's good). I am using those context for the sake of code clarity and reuse, and for

"Request For Comments on a Feature Suggestion" -- just wondering if others would find this useful. Frequently, when something really doesn't make sense, I like to bump up the console logging by editing logger.conf and changing console = error to console = debug,dtmf,error,event,info,notice,verbose,warning and entering logger reload When done, I

Dear all i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is possible to configure a scenario like this: 1) receive a call and put it on-hold in a queue (OK) 2) monitor the queue and trigger an outbound call to a remote number using AMI, setting the channel of the on-hold on a specific var named channel2Link (OK) 3) when the remote number answer, trigger an

Hello, When reloading a diaplan (asterisk 1.6.1.X), I can see in console : [Apr 2 09:02:00] WARNING[2217]: ael/pval.c:2522 check_pval_item: Warning: file /etc/asterisk/extensions.ael, line 621-621: application call to Gosub affects flow of control, and needs to be re-written using AEL if, while, goto, etc. keywords instead! What is then the recommended substitution for Gosub() application

Dear list, today I installed a new asterisk machine, bound to replace my current pbx. I am using a Fritz ISDN card; on the old machine with the drivers coming together with the super-old rpm asterisk installation of SUSE 9.2. The new machine is finally on asterisk 1.0.9, with chan_capi 0.5.4; now I am doing a nightly test. Apparently I can receive calls, but I can't dial out. I seem to

Does IAXy2 have any internal diaplan? What I need to do is have a phone that when picked up automatically dials out in a hotline fashion. I know that I can setup Sipura devices to do this, but I'd rather use IAX2. Alternatively, do any of the cheapish SIP phone support this sort of internal dialplan functionality? Thanks, Michael -- Michael Graves