similar to: 200 OK however still rinnging

Hello Everyone, We are looking for a simple open source auto dialer with "polling" capabilities. What we would like is a program that we can upload leads to, and have asterisk: i) Dial numbers ii) Play pre-recorded iii) If user presses one, forward the call to an agent There are so many solutions out there it's hard to make a decision on what works, what has just a limited free

Hello Everyone, I have a strange issue where the caller's phone keeps ringing even after the 200 OK. I am using the latest version of Asterisk 1.8, and wanted to know if anyone could give me any pointers before posting the SIP debug messages. Kind Regards, Nick from Toronto.

> Many thanks Julian. Are you looking for the pinout for a single plug 2.5mm (cellphone) headset or a dual plug 3.5mm (computer) headset? -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeel<at>jafferali.net

Hi guys, My asterisk didn't send ACK for 200 ok message just for one specific extension. The ATA used by this extension is used by other extensions, with same firmware version. Looking in wireshark, I saw that ATA sent 200ok and asterisk didn't confirm it with ACK. The ATA did this during 20s, after this, asterisk hangup the call. This issue happen only one asterisk start a call.

Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording

I'm configuring some monitoring for a particular java/tomcat application. We have noticed the occasional "Cannot allocate memory" error. When this occurs apache still seems to return a "200 OK" status code. Anyone know how to configure this so that when java has an error, apache will also return some kind of error?

Hello There, I would like to play a busy tone (ie BEEP BEEP BEEP) when the underline carrier sends back 486 Busy Here. Looking at Dial parameters ( http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial), it mentioned something about the r parameter as not being very professional or something like that... Then there was: U(x): Executes, via gosub, routine x on the called channel. This is similar

Hello Everyone, I am required to write a java program that will get our asterisk to: * Query the database for phone numbers * Loop through numbers and dial * Play message * Get dial pressed response - If 1 = Yes - If 2 = No - If 3 = Connect to Agent * AMD Capable * Disposition I am proficient with Java and found the Asterisk-Java API. My questions are: * What is the

We are using an MGCP configuration. There seems to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is how our Vendor sees it: Here's what I see. 1. The first call is initiated. (CRCX) The interesting thing here is that the CA (Call Agent) tells us to go directly into sendrecv mode which means that we start streaming audio immediately. All other CAs that

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

Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

OK try it now - I checked in a few more fixes. On Tue, Feb 22, 2011 at 8:29 AM, Chuck Zhao <czhao at eecg.toronto.edu> wrote: > I still can't build LLBrowse on my Debian5-i386 machine today, > The following is a full build console output. > I am using LLVM-2.8 release build, with needed wxWidgets and CMake. > > Thank you > > Chuck > > sideshow.eecg>time

I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible

Hey, This is probably a rather stilly question... If I pick up my SIP phone that's registered to my asterisk server and dial a number that asterisk recognises as destined for a SIP trunk (could be a static route, or an ENUM lookup) or another SIP device registered on said asterisk server (internal extension to extension call), what route does the actual audio take? The control

Does anyone know where I can get replacement handsets for the Polycom SoundPoint IP phones? Or does anyone have any they want to sell? From the looks of it you have to buy a whole new phone to get a new handset. My vendor, TriaTechCOA, told me I had to buy a whole new phone to get a handset, which is pretty ridiculous. Maybe there is a more sane vendor I should be buying from? Thanks, -Ryan

Hi, Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support roaming and handover and are these functions transparent for handset (then, these functions are implemented in DECT base stations) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080110/4254f602/attachment.htm

Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite

Maybe I'm loosing my mind but I've just noticed that if I put a call on speakerphone and I press speakerphone again it hangs up the call, you would expect it to take the call off speaker back on to the hand piece. I'm using V 1.0.5.22 firmware. Is there any other way to turn off speakerphone I'm missing? Cheers, Dean -------------- next part -------------- An

Hi all, Can someone give me a simple rational explanation why a $5 analog handset gives me no echo whatsoever on an analog PSTN line, but PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require software-based echo cancellation. Surely a $5 analog handset does not have an "echo canceller". The echo I mean is when I hear myself while talking to another party. I have heard

While I'm asking about the Polycom ip500, the answers for all phones where mic/handset/headset levels are adjustable would be of interest to many I'm sure. For the ip500, the default value for the handset seems to be voice.gain.tx.analog.handset="3" I've noticed that echo all but goes away when one reduces the mic volume on almost any phone. My question is, for you users