similar to: problem about TDM400P ringback detection

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Great R people, I have noticed a strange behaviour in read.delim() and friends in the R 2.7.0 version. I will describe you the problem and also the solution I already found, just to be sure it is an expected behaviour and also to tell people, who may experience the same difficulty, a way to overcome it. And also to see if it is a proper behaviour or maybe a correction is needed. Here is the

Hi all, I am new to this mailing list and I apologize if my question is naive but I couldn''t find help about it on the CRAN site. I have just manually set a "cran.repo" for yum but I don''t know how can I install the gpg key for the packages. Can you please tell me how to do it? By now I just used the --nogpgcheck and it worked fine. Thank you in advance, margherita

I have a problem using option -n (to test if a sync operation will be done) and I add empty directory in the source tree. If I make: # mkdir source/empty # rsync -n -v -a ... source/ dest rsync does not notes empty dir, but if I remove -n option it synchronize correctly also the empty dir. Why ? I think this is important because, as I do, a command like `rsync -n' could be inserted in

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Chris Abel writes: >Hello everyone! > >I've had this problem for a while and cant figure it out. When an outside >caller calls an extension on my asterisk system, they do not hear any >sort of ringing. Inside extensions calling other extensions do hear >ringing. We have 3 other asterisk systems that are configured the same >way, but do not have this problem. We think it

Hi, I have PRI configured and up but when i am dialing outside i am not getting any ringback tone but my call is connected. following is my example SIP----------------->PRI ------------> mobile I have set progress=yes in chan_dahdi.conf but still not working if i call inbound from my mobile to internal extension ringing working please help me -S -------------- next

Hello, everyone. Could anyone explain me how does ringback detection works in asterisk. Sometimes, when making a call, my asterisk box doesn't detect a ringback and I just hear silence until the other party picks up the phone. I've checked the SIP messages and they are ok (I'm getting 183 "session in progress"), so I guess I should be debugging the RTP packets. From then on

hi: When iam sending calls through sip a fake ringback tone is generated and then call status can't be viewed (if call is ringing,busy,offline) it just rings and rings. Can i disable this? Thanks in advance. _________________________________________________________________ Windows Live?: Keep your life in sync.

Is it possible to give a false ringbakc on asterisk ? -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -------------------------------------------------------------------------- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or

Hi What command i need to use to make a call with oh323 and hear the ringback sound Thanks

When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no ringback tone when making a call. I have tried adding the 'r' command to the dial sting with no help. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello. I have another issue. When I call in, everything is processed correctly, including voicemail, but I don't hear any ringing/ringback. exten => s,1,Zapateller(answer|nocallerid) exten => s,2,NoOp exten => s,3,Playback(pls-wait-connect-call) exten => s,4,Dial(${PHONE1}&${PHONE2}&${PHONE3}&${PHONE4},15,Ttm) exten => s,5,Answer exten => s,6,Wait(1) exten

I'd like to add a test extension to implement ringback so that I can test a phone's ringer without having to use another channel in another room. The way I'd like to implement this is to dial a test extension, get a tone, hang up, then one second later, have the system call me back at that extension. There is a way to do this which is mentioned in the Asterisk white paper, but it

I am new to the sip side of things and have a question regarding ringback. I don't hear ringback when using the sjphone softphone when dialing internal extensions. It's fine when dialing outside over the pstn. Is this a issue of the softphone, configuration or sip in general? Thank you, -gcc

I would just like to follow-up on the ringback problem I'm getting from *. As I've said in my previous post, I am not hearing the "real ringback" from the Cisco gateway terminating my call. I don't want to provide false ringback from * (r option of dial), because it'll still give me ringback even if I am suppose to hear announcement or fastbusy. Below is captured ISDN

I am noticing that calls that come from our IAX pstn gateway provider and terminate to our Asterisk IVR do not receive ringing when an extension is dialed. For example: 1. An inbound PSTN caller calls our number 2. Asterisk answers and provides greeting 3. PSTN user dials extension of internal SIP phone 4. No ringback is heard from PSTN callers perspective 5. SIP user picks up or the

Hi All We have Grandstream 102's running ver X.18. When hanging up after a call has been made the grandstream seems not to disconnect the call and when you put the handset down the phone rings only to pick it up and be on the same call. This is happening quite often and gets very irritating. Can anyone help with this? Regards Doug

I am trying to get Asterisk to emulate the sounds of the earlier telephone systems, and the settings in [us-old] are pretty helpful. The only thing lacking is ringback tone, which is not quite as complex as the real phone systems of the day. For example, it is true that a ringback tone commonly used is 420Hz modulated by 40Hz. This is what shows up in [us-old]. But that modulated tone was

Not everybody here has voicemail and we have more than one person answering the phones. I would like to know that if Person A transfers to Person B, that after a predefined timeout, it would then transfer the call back to Person A. Right now when a new call comes in, it rings 3 phones. When Person A transfers to Person B, it tries for 15 seconds, then transfers the call back to all 3 people