search for: inmediantly

Hello guys, I have this problem when a call is received in my PBX: (Caller) --> (Redirecting Service) --> (E1 PRI) --> (Asterisk PBX) --> (Internal Phone) Reception works fine, but when conversation finishes and the agent at internal phone hangs up, the call at caller's side is still alive for many seconds until it hangs up. The problem is that Telephone Company is billing me

Hello all, I want to setup Asterisk to forward a call if the dialed extension is busy. I do not want to wait on the line until the extension timeout expired. What I want is when I dial am extension currently Busy (Talking with someone), asterisk inmediately forwards my call to an extension I previosly defined. Someone could help me? Any clue will be appreciated. Regards from Spain. Ismael

HI all, Is it possible to install a virtual guest (FreeBSD and OpenBSD mainly) from kvm host's console? I have tried to do some search about this but I haven't found anything about it. My idea is to launch "virsh start guest1" and inmediately "virsh console guest1", but it doesn't works until serial console is configured inside guest .... Any tip? Thanks.

Am Montag, den 16.03.2020, 15:17 +0000 schrieb Rowland penny via samba: > > After deciphering the above, Sorry. evolution has reformatted my text :-( > it is a known feature, only when a user > logs in can you be sure to get a full list of the users groups. Very strange. I've recreated a new user and put it in all groups and he was inmediately in all groups. What does mean to

We used samba 2.0.6 under Redhat 6.1 with W98/NT4 clients without any problems. Now we upgraded one PC with Windows2000. In the beginning everything worked fine. But suddenly there are problems: Browsing a specific subdirectory of a share on the server cause a inmediate crash of W2k with a automatic restart. I upgraded to Samba2.0.7, again W2k crashes. I connect to this subdirectory from a

Hello, hopefully someone will remember my previous problem to load the Rcmdr library from within GNU R resulting in an error message: "libnvidia-tls.so.1: cannot handle TLS data". Some suggestions have been raised by Christian Schulz and others that unfortunately didn't work around this error. Nevertheless I felt very grateful for your suggestions! This morning I investigated

> We tried that too...but still it returns an empty map. I'm not sure what's going on then. :) I think that Vikram is planning to update CVS soon, you might try that, as there has been significant updates. I'm not sure why getNodeForValue would work for you but getScalarMap doesn't... getNodeForValue USES the scalar map! :) > We also saw that Scalar Type has been

Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm testing a telephone connected to FXS port of a Sangoma A200 card. But I'm observing that callerid is not working. The configuration that I'm using in chan_dahdi.conf is the following one: - --------------------------------------------------------------------- ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit

hi, I still has a few other problem with dovecot and mozilla. - when I open mozilla mail I can only see the toplevel folders in the left handside folder list and when I close the imap folder and open it again (click on the arrow left to the imap server in the folder list) mozilla show me all folders and subfolders correctly (but not at the startup). it can be mozilla's bug, but

> I don't know (don't use ices) but I thought I should inform you.. taking > user requests may cause you to violate the RIAA rules regarding that topic, > depending on how you handle it. Be careful. ;) Again, in Spain there is no rule in this sense... I mean, a normal air-FM/AM station can receive a call for a song and play it inmediately. What happens if this spanish station

...inging: NOTICE[540685]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 received The PSTN recipient's phone rings. The client does not receive the typical intermittent tone/signal that means "the recipient's phone is ringing". When the recipient answers, the call is inmediantly finished. Maybe a short "Hello" can be listened. Asterisk shows a response back from Cisco: Bad Request - 'Invalid IP Address' In sip.conf, Asterisk is forced to use g711ulaw. I've tried other codecs with no success. What is the real problem?. Is it a RTP problem with &q...

Hello: I have found the following problems with outgoing calls with asterisk, compiled with an updated CVS on 22 Oct. 1.- Problem with retries: Whenever I set the MaxRetries parameter, to something greater than 0 in a call-fille, Asterisk ignores the RetryTime parameter and retries every file in the outgoing folder when a new call-file is copied into that folder. So, if I make a call placing

...inging: NOTICE[540685]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 received The PSTN recipient's phone rings. The client does not receive the typical intermittent tone/signal that means "the recipient's phone is ringing". When the recipient answers, the call is inmediantly finished. Maybe a short "Hello" can be listened. Asterisk shows a response back from Cisco: Bad Request - 'Invalid IP Address' In sip.conf, Asterisk is forced to use g711ulaw. I've tried other codecs with no success. What is the real problem?. Is it a RTP problem with &q...

> De: Duncan Sands <baldrick at free.fr> >Para: >Enviado: miércoles, 11 de julio de 2012 9:40 >Asunto: Re: [LLVMdev] dynamic linkage for jit > >Hi Charllls, I didn't really get it. Are you saying that you would execute some >functions using the JIT then, later, those functions will be modified and thus >need to be re-jitted? > >Ciao, Duncan. Hi Duncan,

> Again, in Spain there is no rule in this sense... I mean, a normal > air-FM/AM station can receive a call for a song and play it > inmediately. Yes. In the US as well. But not an INTERNET station. The rules are different if you broadcast over AM/FM than if you broadcast over IP. > What happens if this spanish station broadcasts its content to internet > and a US listener

In a setup I've made i have a problem in the two way origination of the call. Asterisk 1 <==> Public internet <==> NAT <==> Asterisk 2 note the two way pointing arrows. When you call from the *2 to the *1, the call passes, but if you try to call from the *1 to the *2, it will not connect, unless you call from *1 to *2, and inmediately after hang-up, you call in the other

hi all I can not register my new granstream bt100 phone with asterisk, i have old of they working perfectly but they have an older firmware(1.0.5.3). any bady now where i can read about this or now what i have to do??? My sip.conf: [10] type=friend context=unr username=10 callerid=10 usecallerid=yes hidecallerid=no canreinvite=yes host=dynamic dtmfmode=info nat=no mailbox=10 callgroup=1

I've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure what it's supposed to do, but I wouldn't expect it to continue processing the dial plan. Any pointers? Documentation locations that address hanging up would greatly appreciated! TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david@safedatausa.com

Is there a way get the register on the LHS of a llvm instruction? (in case there is one) for example, given a "free %reg773", I want to find the matching "%something = malloc %", where %something is not necessarily %reg773. One way I found was to inmediately follow the use-def chain up from the free to the malloc. But instead, I want to put %reg773 in a set (possibly with many