search for: snom1

...es.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info: sip:192.168.20.1\; answer-after=0) exten => _*2XX,n,Page(${PAGING_DEVICES},dq) The CLI outputs the following: -- Executing AGI("SIP/snom1-b7d0c328", "get-paging-devices.agi|01") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/get-paging-devices.agi -- AGI Script get-paging-devices.agi completed, returning 0 -- Executing GotoIf("SIP/snom1-b7d0c328", "0?i|1") in new stack...

...ls are working. Calling the phone when some other lines are in use on the phone is ok. What does not work though is when the phone is ringing, nobody else can call the phone anymore. That's what * is saying: -- Got SIP response 486 "Busy Here" back from 192.168.1.250 -- SIP/snom1-4a44 is busy

I have a test SIP account set up with WorldCom and I have been trying to have Asterisk register to the WorldCom server with no luck. It appears that the SIP headers are different coming from Asterisk. I have included a packet capture from a successful login with a Windows Messenger client for reference. I have also copied in the SIP packet I captured with sip debug turned on. In my sip.conf file,

...ls are working. Calling the phone when some other lines are in use on the phone is ok. What does not work though is when the phone is ringing, nobody else can call the phone anymore. That's what * is saying: -- Got SIP response 486 "Busy Here" back from 192.168.1.250 -- SIP/snom1-4a44 is busy _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

...0 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ; ;register => me@mysipproxy.org ; Register with a SIP provider ;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as 1234 here. ; [snom1] type=friend host=129.26.10.121 dtmfmode=rfc2833 mailbox=2101 reinvite=no context=local callerid="Studenten 1" <2101> [snom2] type=friend host=129.26.10.122 dtmfmode=rfc2833 mailbox=2102 reinvite=no context=local callerid="Studenten 2" <2102> [snom3] type=friend ho...

Hello, i want to use asterisk as a gateway for H323-Phones. But i cant get it work. I'm using a gatekeeper on another computer. My IP-phone is registered there. Does anybody can sent me an oh323.conf and extension.conf as examples? Thanks in advance Erik Bastian -- NEU : GMX Internet.FreeDSL Ab sofort DSL-Tarif ohne Grundgebühr: http://www.gmx.net/dsl