search for: seraddress

...r the following: *answer call *wait for 4 digit extension *send call to 4-digit extension entered. I tried the following, but that doesn't work... exten => 998,1,Answer() exten => 998,2,Background(agent-newlocation) exten => 998,n,WaitExten(20) exten => 998,n,Dial(SIP/${EXTEN}@${SERADDRESS},60,tr) WaitExten obviously does not fill EXTEN with its value... Anyone any suggestions? Regards, Evert

I have some problem to configure the call from asterisk to ser. [globals] SERADDRESS=xxx.xxx.xxx.xxx:5060 exten => 77,1,Dial(SIP/phonenumbertocall@${SERADDRESS},20,r) Error in Sip Debug ------------------------------------------- NOTICE[25541]: chan_sip.c:6848 handle_response: Failed to authenticate on INVITE to '"Alexg" <sip:sipsourcephone@xxx.xxx.xxx....

>From my extensions.conf: exten => 888,1,Answer() exten => 888,n,WaitExten(20|m) exten => 888,n,Dial(SIP/${EXTEN}@${SERADDRESS},60,tr) This should: * answer * wait 20 seconds for an extension with music on the background * pass the call to that extension on ${SERADDRESS} What am I doing wrong here? I don't even get the background music while WaitExten is active. I doubt that it is active anyway, since I get disconn...

...sk 1.0.9 When I try to make an outgoing call with SIP I get the message " 513 Message too big" back from SER. Any ideas what I am doing wrong? Debug below. SER and Asterisk are running on the same Server SER is on port 5060 Asterisk is on port 5061 In my extension.conf I have the line SERADDRESS=192.219.85.57:5060 in Globals and am using exten =>_5XXX,2,Dial(sip/${EXTEN:1}@${SERADDRESS}) to dial out. Here is the sip debug. -- Executing Ringing("H323/ip$192.219.85.57:2680/5746", "") in new stack -- Executing Dial("H323/ip$192.219.85.57:2680/5746&quot...

...;_9.,1,Goto(default,${EXTEN:1},1) Why am I unable to make outgoing SIP calls? I have also not made any changes to my DNS SVR settings (in case I need to???) I have also tried to have the following alternative in my extensions.conf It is something like this: Asterisk extensions.conf: [globals] SERADDRESS=XXX.XXX.XXX.XXX:5060 [context] exten =>_5XXX,1,Dial(SIP/${EXTEN:1}@${SERADDRESS},20,r) However, this then causes a message "Message to Big" to be displayed in SER. Many thanks for your help. I am probably doing something obvious wrong! Thanks David

...I start my asterisk with the sample of redhat and "/etc/init.d/asterisk restart". Best Regard Charles In my extensions.conf: ------------------------------------------------------------------------------------------------------- [general] static=yes writeprotect=no [globals] SERADDRESS=xxx.xxx.190.248:5060 CISCOADDRESS=xxx.xxx.190.243:5060 [ser] exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@${CISCOADDRESS},60) [cisco5300] exten => 1055,1,Dial(SIP/${EXTEN}@${CISCOADDRESS},60) In my sip.conf: --------------------------------------------------------------------------------------...

>When I call a SIP user, the phone should ring in more than one >extentions. Also more than one phone should be able to register with >asterisk. Right now it is not the case. There is no issue here. You seem to be confused, that's all. A SIP account is a SIP account and an extension is an extension. You can assign an extension to an account (or to multiple accounts) and the tool for

...0005,6,Hangup ;specify context for receiving incoming calls [pstn] ;Note this is just an example there are infinite different ways to handle the incoming call. ;exten => 1234, 1,Wait(1) ;exten => 1234, 2,Playback(beep) ;exten => 1234, 3,Hangup exten => 1234, 1, Dial (SIP/2092@seraddress) ; 1234 is the contact extension, default contact extension is "s" ;exten => 2092,1,Answer() ;exten => 2092,2,Playback(welcome) ;exten => 2092,3,Background(menu) ;exten => 1,1,Dial($316) ;exten => 2,1,Dial($314) [outgoing] ; Dial the Blue Face Speaking Clock exte...

Thank you Melissa. I love the phone but the dial keypad is a little bouncy. I was hoping for a more solid feel like on the analog PT390's or my quality standard, the Nortel 9417CW. Other than the MWI problem, I'd like more documentation on the configuration paramters. I have found little online configuration documentation other than very basic stuff on the Sayson website. I'd

Yo, Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate call divert-feature. This one validates if the extension a call-forward is to be set to is actually valid for the current context and additionally saves this context into the DB and always uses it to originate the divert from, as you can't expect the