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/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; 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/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; 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:[EMAIL PROTECTED]>;tag=i7n7jzxqp3
To: <sip:[EMAIL PROTECTED];user=phone>
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Route: <sip:[EMAIL PROTECTED];user=phone>
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=i7n7jzxqp3
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5d8704ab
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=i7n7jzxqp3
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5d8704ab
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
Route: <sip:[EMAIL PROTECTED];user=phone>
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=i7n7jzxqp3
To: <sip:[EMAIL PROTECTED];user=phone>
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
Route: <sip:[EMAIL PROTECTED];user=phone>
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=i7n7jzxqp3 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5d8704ab Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=i7n7jzxqp3
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5d8704ab
Call-ID: [EMAIL PROTECTED]
CSeq: 2 ACK
Route: <sip:[EMAIL PROTECTED];user=phone>
Contact: <sip:[EMAIL PROTECTED]:5060>
Content-Length: 0
------------------------------------------------------------------------------
--
Anton Yurchenko<[EMAIL PROTECTED]> Digital Generation
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