similar to: Monitoring

Hi, I' ve just connected a carte X100M to my asterisk server running zaptel-1.2.5, libpri-1.2.2 and asterisk-1.2.6 on SUSE 10.0. When I make modprobe wcfxo and modprobe zaptel I haven't any error, I have also chan_zap.so module existing in /usr/lib/asterisk/modules. But, when i run ztcfg, it shows me this: Zaptel Configuration ====================== Channel map: 0 channels configured.

Hello, How many digium cards is supported per asterisk server ? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Hello, Is there an app_snmp for asterisk-1.2.3 ? Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Hello, Asterisk sent back 407 proxy authentication . How can avoid this ? I set insecure=very without success in sip.conf and my sql server . Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Hello, I wish to set a sip uri sip:info@mydomain. I use ser for authorization and authentication (registrar rtpproxy and outbound proxy) I use asterisk 1.2.5 with realtime . the info is used as a hunt group so i add in extension.conf [info] exten => info,1,Answer() exten => info,n,Dial(Sip/84,10) exten => info,n,Dial(Sip/85,10) exten => info,n,Hangup Ser forward sip:info@mydomain

Hi all, look at these lines. I created a queue named info when a caller (extension 86) place a call he is put on queue he sould hear MOH . What's the meaning of : Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]: file.c:821 ast_streamfile: Unable to open 100 (format ulaw): No such file or directory Regards

The list of your interest is R-help not R-sig-mac stefano Il giorno 26/gen/06, alle ore 01:20, Sylvain Charlat ha scritto: > Hi, > > Is there any simple way to get histogram for different levels of > factor? > > Say you have the following data set: > > Island Sp.diam > Moorea 1.21 > Moorea 1.27 > Moorea 1.28 > Moorea 1.22 > Moorea 1.28 > Rurutu

Hi, I have several Polycom 501 connected to asterisk. The phone has an ACD-login function that I'd like to use. But I can't find find much information about this. I've read a post on bugs@digium (http://bugs.digium.com/view.php?id=6119) about this function but I'm not really clear on if this is actually working or not? Has anyone actually used the Polycom ACD-login function

This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com

I have googled this and come up empty. Has anyone had any problems compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am getting when I run make. app_nv_faxdetect.c: In function `nv_detectfax_exec': app_nv_faxdetect.c:210: error: structure has no member named `cid' app_nv_faxdetect.c:227: error: structure has no member named `cid' app_nv_faxdetect.c:265: error:

Hey, whenever I try to load, I get these errors Jan 15 16:37:24 ERROR[7573]: chan_iax2.c:7486 load_module: Unable to bind to 0.0.0.0 port 4569: Address already in use Jan 15 16:37:24 WARNING[7573]: loader.c:345 ast_load_resource: chan_iax2.so: load_module failed, returning -1 == Manager unregistered action IAXpeers == Unregistered channel type 'IAX2' Jan 15 16:37:24 WARNING[7573]:

Hi, I have a big problem with EXEC in AGI scripts: I do, for example, "EXEC Dial SIP/phone1", Asterisk says -- AGI Script Executing Application: (dial) Options: (sip/phone1) Jan 10 14:33:20 WARNING[10567]: chan_sip.c:1389 create_addr: No such host: phone1 Jan 10 14:33:20 NOTICE[10567]: app_dial.c:743 dial_exec: Unable to create channel of type 'sip' I do "EXEC

I can't build it, errors: chan_zap.c:61: #error "You need newer libpri" chan_zap.c: In function `zt_call': chan_zap.c:1806: warning: implicit declaration of function `pri_sr_set_redirecting' chan_zap.c: In function `pri_dchannel': chan_zap.c:7776: structure has no member named `redirectingreason' chan_zap.c:7778: structure has no member named `redirectingreason'

Hi, I still have some trouble with the AGI interface: - I can use EXEC now, but it never gives me the error returned by the executed application, if an error occurs - I can use ANSWER, but I have to put something else behind "ANSWER". If I say "ANSWER", I get "510 Invalid or unknown command". If I say "ANSWER ''" or "ANSWER ." or

Hi, what happens if the dialplan contains something like exten => s,1,AGI(agi://10.0.0.1) exten => s,2,Dial(SIP/phone1|20|tr) etc. - if 10.0.0.1 isn't reachable or doesn't react on the connection? In my test cases, I always got a hangup and no further processing of the dialplan. Any hints? ( the call mustn't go into Nirvana if the AGI server isn't available!) Thanks

Hello. I have just arrived to Asterisk. I would like to know if Asterisk can perform some functionalities I am looking for. I want to allow voip over sip to some users. All of them must have their own user name and password to login to Asterisk so only allowed users can login. All calls started by users have to be redirected to one account at our voip provider. I think those functionalities can

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin