![]() Really destroying SIP dialog 'ZWI3NjVlMzk4YTdlNWQ0YWRmYmQyZDRjOGM5YjQ0Y2Q. Really destroying SIP dialog 'UkrFo5kQBIuQLZqh1xov2g.' Method: REGISTER ![]() Looking for 2 in from-internal (domain 1.1.1.200) ![]() Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon e-event|), combined - 0x1 (telephone-event|) Im trying to specifically get the 7941 to speak with asterisk/freepbx and am unable to get the cnf.xml to load properly, i see the phone request the file via tft and supposedly take the 7,550 bytes but the phone stays unprovisioned. Via: SIP/2.0/UDP 1.1.1.50:5060 branch=z9hG4bK5ffc804cĪuthorization: Digest phone",response="f3f77de0b8da7274b5cb011cc5148b68",nonce="06b8f659",algorithm=MD 5įound audio description format PCMA for ID 8įound audio description format PCMU for ID 0įound audio description format G729 for ID 18įound audio description format L16 for ID 102įound audio description format iLBC for ID 116įound audio description format telephone-event for ID 101Ĭapabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw|g729|slin16|ilbc)/video= (nothing)/text=(nothing), combined - (alaw|ulaw) Scheduling destruction of SIP dialog 50' in 32000 ms (Method: INVITE) WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06b8f659" Via: SIP/2.0/UDP 1.1.1.50:5060 branch=z9hG4bK999aa8c7 received=1.1.1.50Īllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE Specify a different number to store in the phones Received Calls history via. Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-c isco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallbac k,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.5.1Ĭontent-Disposition: session handling=optional Make the phone auto-answer a call based on a dial-plan variable. I am used command “asterisk -r” then used “sip set debug on” and the result show below:
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