similar to: Help on AGI running

Hi * users, I want to know if STUN suport is available with Asterisk. Kindly let me know. I have posted this also in DEV list but none replied to me. thanks, Somesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050808/e26855c9/attachment.htm

Dear all, I have two different data frames, that have two common variables: date and sample. Here is a small extract of both of them > head(traffic) datet sessiont samplet buddleiat 1 07-08-2012 1 1 1 2 07-08-2012 1 1 1 3 07-08-2012 1 1 1 4 07-08-2012 1 2 3 5 07-08-2012 1 2

Hi Folks... Œbeen playing with this for a while, with no luck, so I¹m hoping someone knows it off the top of their head... Difficult to find this nuance in the archives, as so many msgs deal with read.csv! I¹m trying to read a data file with the following structure (a little piece of the actual data, they are actually csv just didn¹t paste with the commas): wavelength SampleA SampleB SampleC

Hi * users, I am only seeing SendText in the available asterisk applications. But I have not seen Receive Text application. I tried on asterisk-1.0.7 and 1.0.9. Can anyone tell me how to use this receive text command. I want to use receivetext command and get text information from an softphone so that that can be routed to some other phone supporting text message. (my soft phones are SIP/IAX

I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found

Hi Asterisk-Users, We have a problem with queues.conf / extensions.conf queues.conf file reads like ... member => SIP/8399 extensions.conf reads like ... exten => 8399, 1, SetCIDNum(${AccountNumber}|a) exten => 8399, 2, Dial(SIP/8399,10,Ttrf) When somebody calls to the queue, we observed that it is not going through extensions.conf (previous two lines) That mean's it is not

Hi All, I have the problem setting up TDM22B card. Steps what I have followed are: [1] compiled zaptel-1.0.9.2 & installed the same. [2] modprobe wcfxo /lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg

Sorry for asking this question again but before I blow 100 dollars on a X100P I need to know this info: So does "SET EXTENSION <new extension>" allow for you to set which extension the rest of the call will occur over? So if a call comes into the switch and I could make the AGI script check the DID or DNIS which is really in the variable agi_dnid? After that I can do a database

Hi, I didn't get any solution in the mailing list. [http://asterisk.linkx.net/asteriskusers/200409/msg01167] What should be the next step? Changing the machine??????? Is it machine dependent?... Regards, Somesh S. Shanbhag --- Tzafrir Cohen <tzafrir@cohens.org.il> wrote: > On Tue, Sep 27, 2005 at 12:13:21AM -0700, somesh s > wrote: > > Hi, > > > > I did

Hi All, I want to do a three-party conferencing using manager api. But I found out from the asterisk-users list that I *MUST* use the meeting room concept. I wanted to know wheather meeting room can be configured dynamically? on the fly? Otherwise, configuring meeting room statically is not scalable. Thanks Regards, Somesh S. Shanbhag --------------------------------- Bring

Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader. Following commands are used - Action: Originate Channel: SIP/111 Application: MeetMe Data: |edwx ActionID: ffe4563 When I use the above, Incoming call will

Hi Folks: I used that one example for AGI script on C web, only to fill the working with the Asterisk. I compiled and it worked great. I executed accidentally the ls -l command in directory where was the source and executable, I noted and was surprised that because the executable size was to further 20 times more than source. I executed the gcc -Os source.c -o executable.agi command several

Guys, Program stack is getting overflowed when I create and destroy encoder instance continuously for around 30 times. How can I ensure stack is completely freed when I destroy the encoder instance? I tried with VAR_ARRAYS. But Its giving me errors while I try to compile. I suppose VAR_ARRAYS allocates local variables as arrays so that they could go on to stack and finally when we leave the

Hi Asterisk-users, I am working with Aterisk Manager API's. I can login successfuly with the following. char buff[256]; strcpy(buff, "Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n"); send(msock, buff, 255); Now I want to try Action: Originate, therefore I tried the following char buff1[256]; strcpy(buff1, "Action: Originate\r\nChannel:

Hi List, Can any one please let me know how to pass arguments to the agi script from the dialplan? I read that it is possible to pass arguments to an AGI script here, http://home.cogeco.ca/~camstuff/agi.html, by entering the variable followed by a vertical bar but it doesn't seem to work for me. I'm using a basic AGI script to query a database and then returns to specific contexts

All, please let me know is there any new version of theora encoder after 1.0beta2. Regards, Nallam. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or

i've just spent the pass 2 days trying to get AGI to work with PHP; i made a lot of silly mistakes along the way which could have been avoided if only there were some kinda howto or samples. at the risk of looking stupid, i decided to shared my experience in hopes that it might help some newbie get going with PHP. 1. first order of business is to be aware of your php environment; i m NOT

When I read variables in AGI scripts, I see only the follwing 13 variables agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode beside these, I found following variables documented on several sites. agi_calleridname agi_callingpres agi_callingani2 agi_callington agi_callingtns Where can I

Dear R-Helpers, I am looking for ways to assess quality of a predictor of variance of a random variable. Here a two related, but yet distinct, setups. 1. I observe y_t, t=1,...,T which is normally distributed with unknown variance v_t (note that the variance is time-dependent). I have two "predictors" for v_t, dubbed v1_t and v2_t, and I want to tell which predictor is better. Here

Is there a way of getting the dialled number from an AGI? Is it passed in the initial variables, or can it be pulled out or passed across from the dial plan? Cheers, Ben Merrills Griffin Internet -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040630/71e0bcda/attachment.htm