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...caller (keypad dialling, E-mail alias typing, etc.) to the receipt by the caller of a meaningful progress information In other words: Call setup time = time from call start (caller press the # key - start the invite) to session progress (183) or ringing (180) Thanks and best regards, Carlo Dimaggio

...001 [user001-sip-in] context=default defaultuser=user001 fromuser=001 fromdomain=sip.xxx.it type=user insecure=port,invite secret=pass001 qualify=yes port=5060 nat=no host=sip.xxx.it canreinvite=no --- extensions.conf [default] exten => user001,1,Noop(Inbound call) Thanks and regards Carlo Dimaggio

Hi all, In order to use the "open url" function of zoiper (it opens an url based on the asterisk $callerid(dnid)), I need rewriting of the dnid. In my dialplan I have: exten => 1000,3,Set(CALLERID(dnid)=newdnid) exten => 1000,4,Noop(${CALLERID(dnid)}) exten => 1000,5,Queue(test-queue) but the callerid(dnid) shows the extension called (the member of the test-queue) and not

================================================================== "Your brain commands your body to "Run forward! Bend! Scoop up the ball! Peg it to the infield!" Then your body says, "Who, me?" " Joe DiMaggio ================================================================== Release Announcements ===================== This is the latest stable release of Samba 3.5. Major enhancements in Samba 3.5.4 include: o Fix smbd crash when sambaLMPassword and sambaNTPassword entries missing from ldap (b...

================================================================== "Your brain commands your body to "Run forward! Bend! Scoop up the ball! Peg it to the infield!" Then your body says, "Who, me?" " Joe DiMaggio ================================================================== Release Announcements ===================== This is the latest stable release of Samba 3.5. Major enhancements in Samba 3.5.4 include: o Fix smbd crash when sambaLMPassword and sambaNTPassword entries missing from ldap (b...

Hi, we have and Asterisk server connected to a Patton Smartnode 4638 with 4 BRI. We configured 4 SIP account on Patton (1001, 1002, 1003, 1004). The system is fully functional, but we have a problem to recognize incoming calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton) or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming from SIP/1004. I have contacted Patton