similar to: calls are getting disconnected automatically

hi, how to write a advanced dial plan for example: dial to a extension(123).if the user didnot pick the call, caller should get a ivr script(Enter 1 to to dial operator and 2 to go to voicemail) If caller press 1 it should dial to the operator,else if he dials 2 it should go to the voicemail of calle's extension. thanks sandeep. -------------- next part -------------- An HTML attachment

Hi All, This isn't really a question, but it's an issue I experienced that was driving me crazy for a few days, so I thought it might be good for the archives. Basically what was happening was everytime a particular customer called (long distance), the line would disconnect immediately after answering. I thought it might have been the phone, so I swapped the phone with another - still

Hi Guys I have an issue where a call is picked up from a queue. The caller asks the person who answered to attended transfer to extension 3082 (for argument's sake.) 3082 picks up the attended transfer and speaks with the outside caller picked up initially from the queue. A few seconds after 3082 has started speaking to the outside caller - 3082's call goes dead in their

Hi, My requirement is something like as given below, a.c => a.obj contains a1() and a2() function b.c => b.obj contains b1() and b2() function main.c => main.obj call to a1, a2, b1, b2 Now, I want to move a1(), a2() from a.obj to b2.obj and on top of function b1() When I call b1() from main, it should call first a1, a2 and then function definition of b1 Can you please give me some

Dear, i using this scenario. jitsi---> asterisk---->EPABX------> Local Telephone when i am calling from jitsi to no 88 its giving this message and getting busy tone. == Using SIP RTP CoS mark 5 -- Executing [88 at myphones:1] Dial("SIP/sandeep-00000004", "DAHDI/g0/88,20,rt") in new stack -- Called g0/88 [Nov 15 09:53:54] WARNING[3169]: chan_dahdi.c:7536

Hi, I have a small call center with 4 Zap lines and 4 agents. Agents login using sip phones with AgentCallbackLogin. I occasionally gets a complaint that when customers call the call center, after the initial greeting is over the call gets cut after playing the thank you message. I started investigating and found that that happens when the call gets transferred to an agent who is making an

hi all, how to set the caller id facility for the TDM400p card in INDIA. thanks sandeep.s -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080118/aa78a793/attachment.htm

It means, AMI application is no more running or crashed or lost network connection with asterisk server. In such cases call is neither answered nor disconnected by Asterisk. I want to detect such state and jump to next dial plan to answer or reject the calls Regards Amit Patkar On September 20, 2016 8:07:23 PM GMT+05:30, Matthew Jordan <mjordan at digium.com> wrote: >On Sat, Sep 17,

Hi, Running Asterisk 1.2.12.1 when dialing a known disconnected number the calls just rings and rings. We never get the "The number you are trying to reach...". If we dial the same number from an Asterisk 1.0.11 server again over PRI, we get the message on the 1st ring. Here is the PRI debug of such a call that just rings and rings. Any ideas? PRI debug sur CPL: -- Executing

Hi, Yes I am looking for IR pass that will do insert call of functions that defined in another file. Links/suggestions that guide me to start for adding IR pass will help me so much. Regards, Deepika On Wed, Feb 10, 2016 at 1:03 PM, mats petersson <mats at planetcatfish.com> wrote: > So how do you know what you want to modify (conceptually)? > > Have you got a IR pass that you

No kidding, every time. I know I have the config via tftp working. Funny story - I was getting nowhere with it and then decided to tcpdump on the tftpd box, and wow! The UIP-200 tftp client was looking for the uniden<mac>.txt in lower-case! Hah! That was easy to fix. Now the config is transferred to the UIP-200 at startup. It registers to the * server. The phone displays time and

All, When calling an invalid number using, I expect to hear: "dooh-deeh-daah We're sorry you have reached a number which has been disconnected ..." And that is indeed what I hear when I dial out from [*] using analog FXO, or VoicePulse or NuPhone. When I dial that same number trough the T1 / PRI interface however, I continually hear ringing, and then the call gets hungup. Any ideas

Hey All, Sometimes we are running into issues with calls randomly being disconnected. We have a couple of phones is in a remote office. We have a PIX-to-PIX VPN setup between the offices. I know neither of these devices provide QOS but we usually don't run into any problems. I notice this problem when we are in a conference call with one of the employees at the remote office and a

Hello, I know there's a problem with Asterisk at the moment in that while it's easy for one caller to dial another (using the dial command), it's tricky to connect two calls that are already in progress. I've been using MeetMe to achieve this (with each caller's call being directed to a dynamically created conference room programatically), and this is working - kind of -

Thank you so much for the reply, Tim ! I have some follow up questions (in the inline replies), it would be great if you can answer those. On Tue, May 27, 2014 at 1:36 PM, Tim Northover <t.p.northover at gmail.com> wrote: > Hi Sandeep, > > On 27 May 2014 02:58, Sandeep K Chaudhary <babbusandy2006 at gmail.com> wrote: >> My aim is to get the bitcode for these programs

Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the cisco phone it then disconnects the call at the same time on the cell I hear 4 beeps and about 5 secs later the line on the cell drops, as anyone

Hi guys, I am compiling some programs with clang which use a couple of libraries. When I compile with the command - "clang -O2 -I. -o hello -ldl -llttng-ust hello.c tp.c", it goes successful. My aim is to get the bitcode for these programs but if I try to generate the bitcode with -emit-llvm option, it produces linking issues as follows : (also at - http://pastebin.com/zkmL2SAH)

Quick update on my issues, Voicemail doesn't pickup also. It just drops the line.. Thank you Chris Tuska ------------------------------ Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the

Dear All, I am using samba 3.0.33-3.7.el5 with openldap for centralized authentication and Nfs mount partition for storage drives. Now the user who logged in the domain and using a shared drive, if they stay ideal for approx 30min. The drive get disconnected and when they try to open it 2-3 times it start working again. Please revert if any buddy else has faced the same problem. Regards, Nitin

I am trying to automatically detect disconnected numbers when using the outbound dialer I have written. * Some numbers hang up immediately with a Cause Code > 0 and no voice treatment * Some numbers get voice treatment with a PROGRESS indication and an associated Cause Code > 0 * Some numbers get voice treatment with a PROGRESS indication and no associated cause code (CC=0) My application