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I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have

Hi! I have a new problem with my SIP device.I have done some changes and now I receive continuosly a SIP message: "501" "Not impelmented" back from the SIP Gateway. I can see that it doesn't register to Asterisk. I have in the SIP device: Registrar 1: UnRegistered to: 2222 registrar: 188.208.12.237 5060 expires: 2000 name: gateway passwd: 123 My

Please I need help, I don't know why,almost every time I dial on my dect phones, the dialtone doesn't go off and * doesn't recognise anything!!!! I'm using two dlink voip gateways, MGCP: DL102s. Any ideas? thanks in advance michelle matis ----- Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis!

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

Hi eveybody again! I don't want to be annoying, but if nobody can help me with this, I'll have to desist of working with SIP.I have some questions about SIP, as I wrote in another mail. I have a SIP Gateway and I have two phones (an analog one and a DECT one) conected to it.Also, I have two Dlink dg102s with four phones conected to them. The main problems are two. Calls between the

Hello Thank for adding mail_log plugin in Dovecot POP3/IMAP server. But when I am use "mail_log" in pop3 secction Dovecot getting the following log message: ... Jan 26 15:34:11 rtr dovecot: pop3-login: Login: user=<gk2>, method=PLAIN, rip=8x.xx.xx.xx, lip=8x.xx.xx.xxx Jan 26 15:34:11 rtr dovecot: POP3(gk2): dlopen(/opt/dovecot/v1.0rc19/lib/dovecot/pop3/lib20_mail_log_plugin.so)

Hi list: I'm having the next problem, it appear that the application ChanIsAvail is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS. I add my dialplan and the output to the cli. THanks. In the example i'm dialing from extension SIP/112 My DialPlan Secction: [macro-callonlyiffree] exten => s,1,ChanIsAvail(${ARG1}|s) exten => s,n,NoOp(${AVAILCHAN}) exten

Hi all. I am using the library Microsoft RTC to connect a windows software with * in a local network via SIP. In my sip.conf I have defined a 'friend' for this software with their username and their password. I don't want that nobody can be connected without the password. When I call to * it says: "Proxy Authentication Required" in the SIP debug. Please somebody can

<-----------------------------------------------------------------------> Changes to user passwords are captured by a special DLL, which traps and then stores the password changes in encrypted form in a private area. On each synchronization schedule, the synchronization service first examines the SAM file for changes, and then checks this private area for passwords to be synchronized. Once