search for: webrainstorm

...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.

...party is still ringing. Our billing software is billing the user (and the carrier is billing us) even with unsuccessful calls. This way we can start billing when the user press the DTMF sequence to unlock audio (even if carriers bill us wrongly) Someone wants to help ?? Regards Edoardo Serra WeBRainstorm S.r.l.

...Used codec is mostly G729 Sometimes on asterisk cli i see some messages like "Avoided initial deadlock for '0x9fd130', 10 retries!" I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l.

...#39; == Starting SIP/180-108c94b0 at call-third-party,s,2 still failed so falling back to context 'default' Do you have some idea to achieve this kind of result ? Maybe I can use ChannelRedirect from Asterisk 1.4 ? Cna you give me a hint on that ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Fo? 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275

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:

...ng SIP/180-108c94b0 at call-third-party,s,2 failed so falling back to exten 's' == Starting SIP/180-108c94b0 at call-third-party,s,2 still failed so falling back to context 'default' Do you have some idea to achieve this kind of result ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l.

...wouldn't like that o happen again because of a slow query or sloq execution of a perl script (it could happen for a lot of reasons) Someone can help with that ? Sorry for the crosspost but I think also asterisk-devel could be involved in it. Tnn in advance for help Regards Edoardo Serra WeBRainstorm S.r.l.

...t have any feedback yet I have an OpenSER acting as a load balancer for 2 asterisks but I don't think it could be responsible for that (I'm not using any kind of RTP proxy, rtp stream goes directly from user to asterisks) Every kind of help is really appreciated Regards Edoardo Serra WeBRainstorm S.r.l.

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 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:

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

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