Cisco 7911g | |
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I have some oddness with this phone.
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+configuration+files+for+SIP Any help on this issue would be really appreciated.
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:7469@xxxxxxx>;tag=001906af068d0002cce99518-da3b5d4e To: <sip:7469@xxxxxxx> Call-ID: 001906af-068d0002-c7a94e00-36e13b76@xxxxxxx CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY Supported: replaces Contact: <sip:7469@xxxxxxx> 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:7469@xxxxxxx>;tag=001906af068d0002cce99518-da3b5d4e To: <sip:7469@xxxxxxx>;tag=as73be6a56 Call-ID: 001906af-068d0002-c7a94e00-36e13b76@xxxxxxx 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
'001906af-068d0002-c7a94e00-36e13b76@xxxxxxx' 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:7469@xxxxxxx>;tag=001906af068d0002cce99518-da3b5d4e To: <sip:7469@xxxxxxx> Call-ID: 001906af-068d0002-c7a94e00-36e13b76@xxxxxxx CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY Supported: replaces Contact: <sip:7469@xxxxxxx> 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:7469@xxxxxxx>;tag=001906af068d0002cce99518-da3b5d4e To: <sip:7469@xxxxxxx>;tag=as73be6a56 Call-ID: 001906af-068d0002-c7a94e00-36e13b76@xxxxxxx CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY Supported: replaces Expires: 3600 Contact: <sip:Test%20Phone@xxxxxxx:5060;transport=udp>;expires=3600 Date: Tue, 30 Sep 2008 10:36:03 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog
'001906af-068d0002-c7a94e00-36e13b76@xxxxxxx' 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:7469@xxxxxxxxxxxxx>;tag=001906af068d0003b2933d3d-4748497c To: <sip:7408@xxxxxxxxxxxxx>;tag=as7cfbb8f0 Call-ID: 001906af-068d0003-87697b57-e8af4b4e@xxxxxxx 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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