search for: inmediat

Hello all I need your support in a big decision in front of two alternatives related with *. I must buy an E1, in order to manage 30 channels, given this big price; or I could opt for a 15 BRIs without cost to replace the same number of channels, and the question that inmediatly emerge is : ? can asterisk manage 15 BRIs ? If yes to the latter, could posible somo guide, for instance, wich Digium hardware I must choose (Card and channel bank) ? How can i configurate it?? Thanks a lot for your attention Jorge Cardona

hello list, I subscribe to Sixtel.net for a DID just to see how it worked. they say DID active inmediatly , but after 4 days , I have no DID , I tried to call tech support office and it seems they all ways are in hollidays, a support ticket no good luck, I call back service also not good luck. so anyone here has experience with them? are they a SCAM? fortunally only put there $10 bucks but , inf...

...nstall and conigure the last version of openssh and it is working like a root user ? $ ssh root at server ? and it?s just fine, even i can use CVS software, but if a try to use it with other user it doesn`t work ? $ ssh any at server ? ask me for a password and it wrote me, ?have a lot of fun? and inmediatly ?connection to server closed? could you please help me. ? ? Enrique Quintanilla Gzz. Auto Summit Commercial Services, S.A. de C.V. Tel. (81) 88 65 73 89 ? ?

...I'm running Asterisk 1.6.1.0 compiled from scratch under debian Lenny and I can't delete message from VoiceMailMain using option 7 Default folder is /var/spool/asterisk/voicemail and it's owned by asterisk:asterisk with 777 permissions Apparently VoicemailMain delete the message and inmediatly undelete it ! This the same issue as in this post : http://www.mail-archive.com/asterisk-users at lists.digium.com/msg198336.html Note that I'm using the spanish voiceset. == Parsing '/var/spool/asterisk/voicemail/from-internal/300/Old/msg0000.txt': == Found -- Playing '/va...

...red but less. It became then clear that nscd was also a problem. The daemon caused the problem: the processlist showed that several instances of nscd were running. But one of the daemons locked the system: it was not possible to fork a new process. Stopping the nscd caused a locked server to run inmediatly as it should and user were inmediatly able to work. On the four locations the nscd is now stopped for one week and there are no problems anymore. For us it is 100% evident that the nscd is a problem, because starting the daemon again, sooner or later the server will stop responding (no new pr...

Greetings all. Some small question regarding schedule poll operation hypercall. 1. struct sched_poll poll.timeout is measured in what unit of time? Secs, ms? ns? 2. After issuing the hypercall_sched_op(SCHEDOP_poll, &poll); if no timeout is used in poll struct how long will I yield the CPU? 3. If I issue the hypercall and the event never comes is it possible to to yield the CPU for ever?

...this error is unrecoverable. After that, I've installed Times32.exe, but still appearing this message. And the last of, I've tried to install a game from Windows 95/98 (works with xp), called PIRATAS, who needs to install quick time to wathc videos, but the window of quick time appears and inmediatly dissapears when installing. Lots of thanks for the help, and by the way... Do you know how to install Diablo 2 in Wine ?

...0 secret=xxxxx disallow=all allow=g723 allow=ulaw host=dynamic canreinvite=no qualify=300 dtmfmode=rfc2833 and this extension.conf [sip] exten => _0114XXXXXXX,1,Dial(SIP/${EXTEN}@xx.xx.xx.xx:5060) where xx.xx.xx.xx is GW ip address but when I place a call from ATA to GW, telephone rings and inmediatly hangs when person answer the phone. When I use only ULAW, all works OK. somebody can tell what I am missing?. someone can help configuring * to use G723 pass through -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-us...

...f SIP users, 01 and 02, with their corresponding mailboxes. My idea is that, if somebody dial these valid extensions, if the users is not connected at the time, a message can be recorded, and also, if somebody dials an extension that matches the pattern, but it's invalid (e.g., 10), the call is inmediatly terminated. The dialplan showed fails, because if the valid SIP user is not registered, the DIALSTATUS is always CHANUNAVAIL, the same that if the extension is invalid, so no call is recorded for my valid user :( Is there a way to do this without having to declare explicitly every valid extensi...

I have a Sipura 2000 working fine, but whenever I dial any extension there is a delay of 5-10 seconds before it starts ringing. However, if I dial the extension and hit the pound key after the number, it goes through right away. Is there any way around this?

Hi there, im having some troubles with my asterisk service, sometimes when im trying to make an outbound call, to any of the phones configured on the asterisk box, it enters inmediatly to voicemail and then hungs up. After that its necessary to stop the service and putting up again manually. Here is a piece of my log file when a call is trying to incoming: "Jun 9 06:30:16 NOTICE[1125329728]: chan_sip.c:4879 handle_response: Peer '1366' is now REACHABLE! Jun...

is the hint 99,hint,ZAP/1 supposed to work or how do I get the lights on the phones to display channels in use in addition to extensions in use?

...Renta4 Equities Paseo de la Habana, 74 28036 Madrid | Spain <http://www.renta4.com/> cid:image001.gif en 01CD0CD2.80C82C00 La información contenida en este mensaje y los documentos adjuntos es confidencial y puede ser privilegiada. Si usted no es el destinatario por favor notifique inmediatamente al remitente, absteniéndose de comunicar el contenido o reproducirlo por cualquier medio. The information contained in this e-mail and any attachments are confidential and may also be privileged. If you are not the named recipient, please notify the sender inmediatly and do not disclose the...

The thing I dislike the most about the 79xx phones is that in DHCP mode, they expect the DHCP server to tell them their TFTP server address. They won't let you set it manually. So if I don't have DHCP server that gives TFTP server info, which is most of the DHCP servers at out there, then the phone won't be able to download any updates made to the SIP000*.cnf file. Using dhcpd on

Hello, I would like to create a function that gets passed a class name and then calls setClass, and a few other functions, inside. I have done this in the past with setmethod, creating accessors for all slots in a set of S4 classes. But setClass is choking when my function is called isnide a package, telling about an error in exists(cname, where). I assume this to be a problem with the

Hello, I currently have a setup on my system with what I call "magic folders" to enable spam filter training. Here's how it works: 1. If you have a false-negative, put the spam into the Spam.Report folder 2. If you have a false-positive (which has all kinds of ugly spamassassin protective markup in it), put the message into the Spam.NotSpam folder

...rk Asterisk as SIP Client and a Huawei softswitch as SIP server. I already got my asterisk registered to the Huawei. Im working with a Sipura SPA 2000 registered to Asterisk. When im trying to make an incoming call from the Huawei to asterisk it rings but when i answered the call drp down inmediatly. The sip debug finally show this message: Reason: Q.850;cause=100;text="Invalid information element contents" And when im trying to make an outgoing call i get the following: SIP/2.0 503 Service Unavailable Reason: Q.850;cause=100;text="Message not compatible with ca...