similar to: CallerID masking

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.

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:

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

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

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 I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 all, What's the best method to send automon call recordings (*1) to the voicemail box of the Asterisk user? Do you have to trap hangups, etc, or is there some global variable that can be set? Thanks! S.

Guys. How are filenames determined for automon and queue recordings enabled on queues.conf? I see the names have some tomestamps or something but is there a way to predefine the filenames to use? Thx!

Hi list, The default file name format for touch monitor (automon) recordings is: auto-${EPOCH}-caller-calee It's possible to use the ${TOUCH_MONITOR} variable to change the 'caller-calee' part, but what about the 'auto-${EPOCH}-' part? I've been trying to use ${MONITOR_EXEC_ARGS} to add some more commands after the somix sequence for mp3 conversion. This should

With our current setup, we have an older avaya system which is linked with our asterisk system via a em wink connection. When you press "2" on the avaya network, it will jump to our asterisk box and then sends DTMF digits. Asterisk listens for those numbers and then responses as soon as it has a match. The problem is with having a "send to voicemail" option. Right now, a user

We have the caller id with name option enabled with our provider, however, our polycom 501 phones will only display the number of the incoming call. Is there a way to see the callerid name from the cli when the call is coming in (like a print in the dial plan)? I'm not sure if the problem is with asterisk or our phones. I did turn on the calleridpres option in zapata, but I'm unsure what

Hello all, I was wondering if anyone has had much experience with Realtime Asterisk. I like the ability to setup my extensions and voicemail boxes in MySQL, but I have a huge worry. What if MySQL crashes. I played with rtcachefriends, but can't seem to find a way to have asterisk store the extension information to ensure the phones will continue to work even if MySQL has a hiccup. Any

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 --

So I might have a bit of a more narrow question from my earlier one. Previous, I had been wondering what would cause a phone dialing into a DID that connects to the asterisk box to get a fast busy. I've noticed the following message: chan_zap.c: Ring requested on unconfigured channel 0/1 span 2 Any idea what would give me this error? And would this cause a fast busy? Thanks again everyone

I am not sure of the best way to do this, so I thought I would query the list. I have about 100 internal extensions ranging from 2000 - 2100. Each internal extension has a external DID number. For example: 2001 = 5552871620. As you can see the internal externsion and DID don't match in any way. What would be the best way to set the DID for when a extension dials out on the PRI? In