similar to: Click to Call

Hi, I made a "follow me" from my phone (111) to my softphone (222)... when someone call me, softphone show my phone number (111).. I?d like to asterisk to forward the calling number, so I could know how is calling me. Thanks a lot. Andrei. _________________________________________________________________ MSN Messenger: converse com os seus amigos online.

Hi everybody,, I need to block incoming collect calls to my Asterisk box but I could not find out where to do that. Went to zaptel.h but I did not see any timing which can be applied to collect calls. Does anybody knows if I can set this up in Asterisk? I'm using an E100P connected to the PSTN and a T100P connected to a Zhone 100. Version: Asterisk CVS-05/30/04-16:28:04 thank you Oz

hello, All, I runa webradio on Icecast and Shoutcast also, but got many compaints of listeners who tell me that they can't listen to the broadcasts on Icecast, because, SOMETIMES, although they already installed Winamp5, when they click on the link, Windows Media player, or Worst: PowerDVD is trying to open the ogg file. Any tips ?? And I'd like to know if there's any Embedded player

Hi, I''ve tried to reproduce the example from section 4.1 (simple source policy routing) but without success. The setup is the same as in the example, with only two differences: there is additionally a gre tunnel which doesn''t seem to disturb here, and one interface does not masquerade at all, as it''s thought to be used from the router only. Besides that, and the

Hi lartc users, When a packet arrived to a network device it was stored on a skb structure and then enqueue on a network stack. One of the tests that I want to do is modify the TCP window size and verify the changes on the bandwidth between two hosts. To do that I need to know how to modify de packet window size and maybe recalculate the checksum value of the TCP packet. All I can do is read

Is there anyone who uses ARS Remedy with wine or CrossOver ??? I can login into the server with username and password but when i use the software, i have a error. It only happens when i have to get some server information. Any help is welcome!!! _________________________________________________________________ MSN Messenger: instale grátis e converse com seus amigos.

Did anybody get this error message before: chan_zap.c:5044 do_monitor: select return -1: Bad file descriptor When it's happening, Asterisk gets freezed and talkers can not hear each other. This message appears like in a loop at the server's screen. thank you Oz _________________________________________________________________ MSN Messenger: instale grátis e converse com seus amigos.

When minimizing, I dont like the icon and would like to know how to change it? __________________________________________________ Converse com seus amigos em tempo real com o Yahoo! Messenger http://br.download.yahoo.com/messenger/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

>From: "Adriano Vasconcelos" <adrianobr2000@hotmail.com> >To: <samba@lists.samba.org> >Subject: samba with Linux and Windows clients >Date: Mon, 19 Jan 2004 14:25:18 -0200 > >I don't autenticate linux in the samba server. The usernames are known but >the passwd are invalid. It happend when try install the pamsmb and NIS. >What I do? > >In

Dihelson, Read my tutorial here: http://www.mcld.co.uk/flatfour/player/about/ I hope it helps! It does exactly what you ask, using Quicktime embedded in a webpage. Our stream is MP3 rather than Ogg but the process should be the same. The only problem is that many people don't have the Ogg codec installed in Quicktime... you'd need to tell them how to install the qt ogg plugin from

Hi, Digium's E400P has RJ45 conector and my E1 link has BNC concetor. Could someone tell me the cable pinouts to make this conection? thanks Eduardo

Hi, I'm use a DSP C5510 to implement a solution with voice, but I'm a beginner in the world of DSP, any one has a example or documents about how to use a speex in DSP's ???? Thanks. Att.; Rafael Vieira Gon?alves skype: rafaelvieira.goncalves msn: tux_surf at hotmail.com email: daconfama at gmail.com P Antes de imprimir, pense em sua responsabilidade e compromisso com o

A SipPulse acaba de liberar o TFPS (www.tfps.co), solu??o para combate a fraudes de fomento de tr?fego internacional em telefonia. O sistema ? capaz de detectar 99.99% das tentaivas de fraude em tempo real. Durante o Webinar, abordaremos como proteger servidores Asterisk e Elastix/FreePBX de fraudes, medidas basicas como configura??o de firewall e remo??o de servi?os desnecess?rios e em seguida

Thanks Jim, But i didnt find this project... It's in CCS folder or on TI site ????? Thk's Em 08/04/2008, ?s 15:52, Jim Crichton escreveu: > The TI directory of the Speex source distribution contains a C5509A > project that builds and runs in TI's Code Composer Studio > simulator. This project does file I/O to files specified in the > main source file. See

Hi list, I'm trying to set up a TDM40B, but modprobe returns the fowling errors: asterisk:~# modprobe wcfxs ZT_CHANCONFIG failed on channel 1: Invalid argument (22) /lib/modules/2.4.18/misc/wcfxs.o: post-install wcfxs failed /lib/modules/2.4.18/misc/wcfxs.o: insmod wcfxs failed Could someone give me a help? Thanks in advance Eduardo

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

https://bugzilla.mindrot.org/show_bug.cgi?id=2547 Bug ID: 2547 Summary: ssh-ext-info: missing server signature algorithms Product: Portable OpenSSH Version: 7.2p1 Hardware: All OS: All Status: NEW Severity: normal Priority: P5 Component: sshd Assignee: unassigned-bugs at

Hi, I have an ansterisk and a cisco 827-4v registered to a Gatekeeper. asterisk has two extensions: exten => 223,1,Dial,OH323/BYEXTENSION@827PD exten => 730,1,Dial(IAX/eduardo@10.0.11.103) (IAX are working well) When I try to call each other, gnugk shows a ARJ: ARJ|10.0.11.112:1720|223:dialedDigits|730:dialedDigits|false|resourceUnavailable I think this could be a codec

Hi, I'm using Dovecot with LDAP. My users have an home directory and a field named 'mailMessageStore'. Then the full mailbox path is 'homeDirectory/mailMessageStore' In dovecot-ldap.conf i configured as follows : user_filter = (&(objectClass=qmailUser)(cn=%u)) user_attrs = qmailUID=uid,qmailGID=gid,homeDirectory=home,mailMessageStore=mail=maildir:~/%$ Everything is

Hi list, Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? Thanks Eduardo