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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 Gil.

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.

...d 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 log in? Is "smbclient //server/share -U user%pw" enough? Connecting "sta" with smbclient has access to all groups. Making su - sta ; id Not all groups will be shown. > > What is interesting are the ID's in the 4000000 range, why this...

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

...your suggestions! This morning I investigated this problem a bit more in depth. It turned out that a buggy NVidia driver (libnvidia-tls.so) is loaded by defualt provoking the reported problem. After renaming /usrlib/tls/ too /usr/lib/tls_old (can also be entirely removed!) the problem vanished inmediately and R Commander works smoothly out of the box! Just thought this might be of interest in case of someone else is affected by this NVidia related problem! regards Thomas

> 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

...Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is being playing so I always have to background a sound to be able to receive DTMF tones. When I press a button (looking for IVR interaction) asterisk catches the correct key but the background song is inmediately muted and the next pulsation is not detected because of the previous problem. Is there a patch or a method to solve this problem? Thanks in advance. Carlos. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachment...

-----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

...ders only sometime some message "disappear"! which means we see in OE6 a red line over the message (as it would be deleted), but no one delete it (yes I know you don't belive me, I also don't belive our users, but once I saw it). just click on it to see the message and inmediately become as deleted. while when I look at in the server the file is still there and _not_ deleted. and OE6 show "the message already removed from the server". - what about the ssl (imaps) problem? yours. -- Levente "Si vis pacem para bellum!&quo...

...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 broadcasts its content to internet and a US listener listens to that station? Would it be violation? If a song is even made by a US artist but the promo CD is given to the spanish station by a record company in spain who bought the rights to control the copies...

Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls directly via Cisco to the PSTN, the call works successfully. If the client softphone calls via Asterisk to other SIP internal extension, it work fine too. The problem is when a client calls an Asterisk extension, and Asterisk transfers

...is copied into that folder. So, if I make a call placing a call-file in the outgoing folder, nobody answers and my call-file stays in the outgoing folder waiting for 'RetryTime' seconds (say 120 seconds), if another call-file is placed in the outgoing folder the previous call is retried inmediately before the RetryTime finishes. 2.- Problem with high-volume calls: When I put lots of call-files in then outgoing folder, from time to time asterisk show an error in Master.csv. The error is OutgoingSpoolFailed, and shows no info about the call, just "failed" as the extension. I move...

I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about

...earn a better alternative the main drawback of that would be, i think, to have to relink all dependencies into the fat module everytime one of the modules change, since as far as i know, you cannot remove symbols already linked in a module and relink a new version Another question that is more inmediately relevant to me (since it potentially affects or invalidates my current implementation plan) is about the transitivity of the mappings between symbols of different modules. Let me explain with a minimal example; say module A implements symbols used in module B, so B is built and when the Call/Invo...

> 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 listens to that station? Would it be violation? That'...

...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 way. I'm pretty sure it's a NAT loosing state too fast, and i can do nothing to fix the NAT. Is there a way to have a keep-alive between the two * boxes? I read about a config in iax.conf called qualify=yes to do that? does the qualify parameter activa...

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 other %things), and when I find "%something = malloc" look for %something in the set. One way to do it (as far as I see) is to store the "defs" in the set. Bu...