similar to: SIP/IAX peers UNREACHABLE and audio loss

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, 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 ChanSpy or

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

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 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 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

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 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

Having trouble running make on res_perl: [root@charlie res_perl]# make perl -MExtUtils::Embed -e xsinit gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/ -I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk \"-DASTVARLIBDIR=\"/var/lib/asterisk\"

Hi, Can anybody tell me which version of res_perl I have to install on Asterisk 1.2.4. I tried to compile res_perl version 3.5 on Asterisk 1.2.4 and I got the following error. gcc -Wall -DRES_PERL_BASE="\"/usr/local/res_perl\"" -DMULTIPLICITY - D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -Wdeclaration-after-statement

Dear List after a lot googling and watching source example of FastAGI i cant find a simple way to convert a very simple perl AGI script... perhaps im not a developer.. Why i have need to use FastAGI?...Very load CPU usage on my box... with only 100 calls.. So i have two way res_perl or FastAGI on some other box.. I cant test res_perl becasue when i try to compile it i have this error:

Latest version of res_perl is up also. http://www.bkw.org/~brian/res_perl.tar.gz Brian Asterlink.com

On a new * asterisk install onto new install Gentoo 2003.4 upon startup of asterisk: WARNING[16384]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/res_perl.so: undefined symbol: PL_thr_key WARNING[16384]: loader.c:429 load_modules: Loading module res_perl.so failed! perl -v = v5.8.5 built for i386-linux-thread-multi <= I installed ithread support in perl Have not been able to

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