I have some oddness with this phone.
The Phone registers with Asterisk (1.4.21), however when I try to make a call it users the default context and not the one that should be applied when it registers. Below are the snippets of the sip.conf and then the debug about the registration. The config on the phone is the default one found @ http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configu ration+files+for+SIP Any help on this issue would be really appreciated. Regards Sean [general] port = 5060 ; Port to bind port bindaddr = 0.0.0.0 ; Address to bind to externip = X.X.X.X ; Address that we're going to put in SIP messages if we're behind a NAT ;localnet = 255.255.255.0 ; Internal NETWORK address ;localmask = 255.255.255.0 ; Internal netmask context = bum ; Default for incoming calls srvlookup = yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 tos=reliability maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=360 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY videosupport=yes ; Turn on support for SIP video ;disallow=all ; Disallow all codecs ;allow=alaw allow=g729 [7469] username=7469 secret=11223344 type=peer context=sip fromuser=7469 host=dynamic nat=no canreinvite=no callerid="Test Phone" <7469> --- REGISTRATION --- <--- Transmitting (no NAT) to x.x.x.x:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKe590c006;received=x.x.x.x From: <sip:[EMAIL PROTECTED]>;tag=001906af068d0002cce99518-da3b5d4e To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 <------------> ast-office*CLI> <--- Transmitting (no NAT) to x.x.x.x:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKe590c006;received=x.x.x.x From: <sip:[EMAIL PROTECTED]>;tag=001906af068d0002cce99518-da3b5d4e To: <sip:[EMAIL PROTECTED]>;tag=as73be6a56 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75af4945" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) Sending to x.x.x.x : 5060 (no NAT) ast-office*CLI> <--- Transmitting (no NAT) to x.x.x.x:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKd526fdb0;received=x.x.x.x From: <sip:[EMAIL PROTECTED]>;tag=001906af068d0002cce99518-da3b5d4e To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 <------------> ast-office*CLI> <--- Transmitting (no NAT) to x.x.x.x:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKd526fdb0;received=x.x.x.x From: <sip:[EMAIL PROTECTED]>;tag=001906af068d0002cce99518-da3b5d4e To: <sip:[EMAIL PROTECTED]>;tag=as73be6a56 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: <sip:[EMAIL PROTECTED]:5060;transport=udp>;expires=3600 Date: Tue, 30 Sep 2008 10:36:03 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) ---- CALL ---- Sending to x.x.x.x : 5060 (no NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 116 Found RTP audio format 101 Peer audio RTP is at port x.x.x.x:32384 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format iLBC for ID 116 Got unsupported a:fmtp in SDP offer Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port x.x.x.x:32384 Looking for 7408 in bum (domain 192.168.1.252) <--- Reliably Transmitting (no NAT) to x.x.x.x:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKbf07a60a;received=x.x.x.x From: "7469" <sip:[EMAIL PROTECTED]>;tag=001906af068d0003b2933d3d-4748497c To: <sip:[EMAIL PROTECTED]>;tag=as7cfbb8f0 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Sep 30 11:37:39] NOTICE[11826]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '7408' rejected because extension not found.
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