[asterisk-users] Cisco 7911g

Tue Sep 30 05:54:19 CDT 2008

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:7469 at x.x.x.x>;tag=001906af068d0002cce99518-da3b5d4e

To: <sip:7469 at x.x.x.x>

Call-ID: 001906af-068d0002-c7a94e00-36e13b76 at x.x.x.x

CSeq: 101 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:7469 at x.x.x.x>

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 at x.x.x.x>;tag=001906af068d0002cce99518-da3b5d4e

To: <sip:7469 at x.x.x.x>;tag=as73be6a56

Call-ID: 001906af-068d0002-c7a94e00-36e13b76 at x.x.x.x

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 at x.x.x.x' 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 at x.x.x.x>;tag=001906af068d0002cce99518-da3b5d4e

To: <sip:7469 at x.x.x.x>

Call-ID: 001906af-068d0002-c7a94e00-36e13b76 at x.x.x.x

CSeq: 102 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:7469 at x.x.x.x>

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 at x.x.x.x>;tag=001906af068d0002cce99518-da3b5d4e

To: <sip:7469 at x.x.x.x>;tag=as73be6a56

Call-ID: 001906af-068d0002-c7a94e00-36e13b76 at x.x.x.x

CSeq: 102 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Expires: 3600

Contact: <sip:Test%20Phone at x.x.x.x: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 at x.x.x.x' 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 at 192.168.1.252>;tag=001906af068d0003b2933d3d-4748497c

To: <sip:7408 at 192.168.1.252>;tag=as7cfbb8f0

Call-ID: 001906af-068d0003-87697b57-e8af4b4e at x.x.x.x

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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