similar to: Queue Status

Hello, When dialling a SIP phone which is already in a call the caller hears a "regular" ringing tone and does not know that the called party is engaged in another call. Is there a supported way inside SIP to tell the calling party to play a stuttered ringing tone? Thanks! __Yehavi:

I would like to know if anyone here knows the answer to the following question I need to implement the following conferencing feature for my agents. 1. Agent receives call from caller 2. Agent conferences a verification service 3. After finishing the verification, agent needs to drop third party (Verification service) and continue on the line with caller. My problem

Hello all, I currently have all outgoing calls set to mask the caller id so it will always appear to be coming from our main number. The problem I'm having though, is with both the call detail in mysql and with the automon (recording) feature. It shows the originating number as the number I masked it to, rather than the actual person calling. How can I go about having both the destination see

I am looking for a way to automatically close a meetme conference when either a user hangs up or through an agi call? Some method that would automatically terminate the meetme. Is there a way to do that? Jerry

All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. Any suggestions? - sf

Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l.

Hi Guys, we're needing a special implementation on Asterisk Our intention is to contribute the development and share back the code to Asterisk community Here is what we need: - An option to Asterisk Dial command which, if used, when calls is answered gives monodirectional audio (Caller should hear the called party but not vice-versa) - A DTMF sequence (maybe handled in features.conf) for

Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently

hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because

Hi guys, I need to realize a sort of automatic call monitoring dialplan. This is exactly what I need: - A user originate a call - When the call is bridged (or just before) I need to invite automatically a third party to the conversation that should hear the audio channel but not speak (it's a monitoring application for a callcenter) The person in charge of monitoring cannot use ChanSpy or

Hi guys, as I wrote in a previous thread I was experiencing dropped audio (apparently randomly) and SIP + IAX peers getting REACHABLE / UNREACHABLE without reason, servers were in the same LAN. Investingating deeply in the problem I also noticed that 'show channels' command on the CLI, sometimes were returning strange results, for example it wasn0t showing some channels I was sure

Hi guys, I need to realize a sort of automatic call monitoring dialplan. This is exactly what I need: - A user originate a call - When the call is bridged (or just before) I need to invite automatically a third party to the conversation that should hear the audio channel but not speak (it's a monitoring application for a callcenter) The person in charge of monitoring cannot use

hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq --

Hi, i''m using to ziya to display some graph. If i use it running mongrel as single instance all work good (very slow but work). If i try to use it with pound as htto proxy ziya charts wait forever for data...that never arrive. If i check the mongrel log it seemes to make queries on the database and retrieve data,but nothin appera. Anyone have an idea on how i can solve this problem?

Dear R users, I would need some help in building R using minGW in windows 8.1. After using the command *configure *(./configure --enable-R-shlib --with-readline=no --with-x=no), I use the command *make. *This results in the following error: [...] make[2]: Leaving directory `/home/Edoardo/r-3.1.2/src/nmath' make[2]: Entering directory `/home/Edoardo/r-3.1.2/src/unix' make[3]: Entering

Hello everyone, I'm in the process of running an Icecast server and I would like to know some best pratices. 1. Should I place Icecast on port 8000 or should I change that to one more common (80, 443...)? 2. Should I place the server behind a webserver like ngingx or apache? 3.Can I disable the login interface? what can be disabled? My best guess is to run icecast behind a webserver,

Hello, I'm trying to build LLVM on the last version of Cygwin, but the 'make install' command terminates with errors. Please find attached the config.log file and the final part of the make output. Can you help me? Thanks ******************************************* Silvano Rivoira Dipartimento di Automatica e Informatica Politecnico di Torino Corso Duca degli Abruzzi 24

Hello *, I''m trying to setup Shorewall under Ubuntu 7.04 and Xen configured to use network-dummy instead of network-bridge (network-bridge seems to be buggy at the moment under Debian/Ubuntu). Is there a shorewall config example I can use in combination with network-dummy? In particular, with network-dummy there is no peth interface and the bridge include the real eth interface. I

Hi all, I have a problem with Asterisk dropping audio. While in call, audio gets dropped for a while (from 5 to 60 secs, and obviously users often hangup, this means that I'm not sure the audio is always coming back after 60 secs), in the meantime the call remains up and no SIP signalation is generated. It happens randomly so it's very difficult to debug. I cannot see common