[Asterisk-Users] Making calls from snom 100
Anton Yurchenko
phila at dg.net.ua
Fri Jun 27 03:30:23 MST 2003
Hello,
I`m trying to make a call from the snom 100( SIP mode) but whatever
number I dial I get a 404 error from Asterisk. Here are my configs and a
dump from "sip debug" . But if I make a call from a Zap line (see
extension 2382031), it rings the snom phone
sip.conf:
------------------------------------------------------------------------------
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
allow = alaw
;disallow = all
;srvlookup = yes ; Enable SRV lookups on outbound calls
;tos=lowdelay
;tos=184
;maxexpirey=3600 ; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;
[100]
context = default ; Default for incoming calls
type=friend
#username=ipphone1
secret=phila
host=dynamic
dtmfmode=inband ; Choices are inband, rfc2833, or info
defaultip=172.22.0.199
;mailbox=100 ; Mailbox for message waiting indicator
[200]
context = default ; Default for incoming calls
type=friend
#username=ipphone2
secret=phila
host=dynamic
dtmfmode=inband ; Choices are inband, rfc2833, or info
defaultip=172.22.0.200
;mailbox=200 ; Mailbox for message waiting indicator
------------------------------------------------------------------------------
extensions.conf
------------------------------------------------------------------------------
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
[default]
exten => 100,1,Dial,SIP/100
exten => 200,1,Dial,SIP/200
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,2,Dial(IAX2/guest at misery.digium.com/s at default) ; Call the Asterisk demo
exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,4,Goto(s,6) ; Return to the start over message.
;exten => 2382031,1,Playback(demo-abouttotry); Let them know what's going on
;exten => 2382031,2,Dial(IAX2/guest at misery.digium.com/s at default) ; Call the Asterisk demo
;exten => 2382031,3,Playback(demo-nogo) ; Couldn't connect to the demo site
;exten => 2382031,4,Goto(s,6) ; Return to the start over message.
exten => 2382031,1,Dial(SIP/100),tTm
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,2,Echo ; Do the echo test
exten => 600,3,Playback(demo-echodone) ; Let them know it's over
exten => 600,4,Goto(s,6) ; Start over
------------------------------------------------------------------------------
the "sip debug" dump:
------------------------------------------------------------------------------
INVITE sip:172.20.0.170 SIP/2.0
Via: SIP/2.0/UDP 172.22.0.199:5060;branch=z9hG4bK-ubuamvlt6h17
Max-Forwards: 70
From: "Anton Yurchenko" <sip:100 at 172.20.0.170>;tag=i7n7jzxqp3
To: <sip:200 at 172.20.0.170;user=phone>
Call-ID: 3c267e34226a-wohzkq5t9qqd at 172.22.0.199
CSeq: 1 INVITE
Route: <sip:200 at 172.20.0.170;user=phone>
Contact: <sip:100 at 172.22.0.199:5060>
User-Agent: snom Version 1.15u
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSA
GE
Supported: timer, 100rel, replaces
Session-Expires: 7200
Content-Type: application/sdp
Content-Length: 257
v=0
o=root 90 90 IN IP4 172.22.0.199
s=SIP Call
c=IN IP4 172.22.0.199
t=0 0
m=audio 10030 RTP/AVP 3 18 0 8 101
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
17 headers, 12 lines
Using latest request as basis request
Sending to 172.22.0.199 : 5060 (non-NAT)
Capabilities: us - 14, them - 270, combined - 14
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.22.0.199:5060;branch=z9hG4bK-ubuamvlt6h17
From: "Anton Yurchenko" <sip:100 at 172.20.0.170>;tag=i7n7jzxqp3
To: <sip:200 at 172.20.0.170;user=phone>;tag=as5d8704ab
Call-ID: 3c267e34226a-wohzkq5t9qqd at 172.22.0.199
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="4bfa4590"
Content-Length: 0
to 172.22.0.199:5060
Sip read: CLI>
ACK sip:172.20.0.170 SIP/2.0
Via: SIP/2.0/UDP 172.22.0.199:5060;branch=z9hG4bK-ubuamvlt6h17
Max-Forwards: 70
From: "Anton Yurchenko" <sip:100 at 172.20.0.170>;tag=i7n7jzxqp3
To: <sip:200 at 172.20.0.170;user=phone>;tag=as5d8704ab
Call-ID: 3c267e34226a-wohzkq5t9qqd at 172.22.0.199
CSeq: 1 ACK
Route: <sip:200 at 172.20.0.170;user=phone>
Contact: <sip:100 at 172.22.0.199:5060>
Content-Length: 0
10 headers, 0 lines
Sip read: CLI>
INVITE sip:172.20.0.170 SIP/2.0
Via: SIP/2.0/UDP 172.22.0.199:5060;branch=z9hG4bK-ly5frg54qsjt
Max-Forwards: 70
From: "Anton Yurchenko" <sip:100 at 172.20.0.170>;tag=i7n7jzxqp3
To: <sip:200 at 172.20.0.170;user=phone>
Call-ID: 3c267e34226a-wohzkq5t9qqd at 172.22.0.199
CSeq: 2 INVITE
Route: <sip:200 at 172.20.0.170;user=phone>
Contact: <sip:100 at 172.22.0.199:5060>
User-Agent: snom Version 1.15u
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSA
GE
Supported: timer, 100rel, replaces
Session-Expires: 7200
Content-Type: application/sdp
Content-Length: 257
Proxy-Authorization: Digest username="100",realm="asterisk",nonce="4bfa4590",uri
="sip:",response="8b80b1d340386d67b378dd73799a8977",algorithm=md5
v=0
o=root 90 90 IN IP4 172.22.0.199
s=SIP Call
c=IN IP4 172.22.0.199
t=0 0
m=audio 10030 RTP/AVP 3 18 0 8 101
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
18 headers, 12 lines
Using latest request as basis request
Sending to 172.22.0.199 : 5060 (non-NAT)
Capabilities: us - 14, them - 270, combined - 14
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 172.20.0.170 in default
Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.22.0.199:5060;branch=z9hG4bK-ly5frg54qsjt
From: "Anton Yurchenko" <sip:100 at 172.20.0.170>;tag=i7n7jzxqp3
To: <sip:200 at 172.20.0.170;user=phone>;tag=as5d8704ab
Call-ID: 3c267e34226a-wohzkq5t9qqd at 172.22.0.199
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:@172.20.0.170>
Content-Length: 0
to 172.22.0.199:5060
Sip read: CLI>
ACK sip:172.20.0.170 SIP/2.0
Via: SIP/2.0/UDP 172.22.0.199:5060;branch=z9hG4bK-ly5frg54qsjt
Max-Forwards: 70
From: "Anton Yurchenko" <sip:100 at 172.20.0.170>;tag=i7n7jzxqp3
To: <sip:200 at 172.20.0.170;user=phone>;tag=as5d8704ab
Call-ID: 3c267e34226a-wohzkq5t9qqd at 172.22.0.199
CSeq: 2 ACK
Route: <sip:200 at 172.20.0.170;user=phone>
Contact: <sip:100 at 172.22.0.199:5060>
Content-Length: 0
------------------------------------------------------------------------------
--
Anton Yurchenko<phila at dg.net.ua>
Digital Generation
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