[Asterisk-Users] Making calls from snom 100

Anton Yurchenko
Fri, 27 Jun 2003 06:36:29 -0000

Anton Yurchenko wrote:

> Hello,

The Issue is fixed by setting in snom100 under Settings->SIP-> Stack
treat as: to address instead of route.

Than happend becouse somebody has been plaing with the phones without me :)



>
> 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@misery.digium.com/s@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@misery.digium.com/s@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@172.20.0.170>;tag=i7n7jzxqp3
> To: <sip:200@172.20.0.170;user=phone>
> Call-ID: 3c267e34226a-wohzkq5t9qqd@172.22.0.199
> CSeq: 1 INVITE
> Route: <sip:200@172.20.0.170;user=phone>
> Contact: <sip:100@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@172.20.0.170>;tag=i7n7jzxqp3
> To: <sip:200@172.20.0.170;user=phone>;tag=as5d8704ab
> Call-ID: 3c267e34226a-wohzkq5t9qqd@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@172.20.0.170>;tag=i7n7jzxqp3
> To: <sip:200@172.20.0.170;user=phone>;tag=as5d8704ab
> Call-ID: 3c267e34226a-wohzkq5t9qqd@172.22.0.199
> CSeq: 1 ACK
> Route: <sip:200@172.20.0.170;user=phone>
> Contact: <sip:100@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@172.20.0.170>;tag=i7n7jzxqp3
> To: <sip:200@172.20.0.170;user=phone>
> Call-ID: 3c267e34226a-wohzkq5t9qqd@172.22.0.199
> CSeq: 2 INVITE
> Route: <sip:200@172.20.0.170;user=phone>
> Contact: <sip:100@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@172.20.0.170>;tag=i7n7jzxqp3
> To: <sip:200@172.20.0.170;user=phone>;tag=as5d8704ab
> Call-ID: 3c267e34226a-wohzkq5t9qqd@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@172.20.0.170>;tag=i7n7jzxqp3
> To: <sip:200@172.20.0.170;user=phone>;tag=as5d8704ab
> Call-ID: 3c267e34226a-wohzkq5t9qqd@172.22.0.199
> CSeq: 2 ACK
> Route: <sip:200@172.20.0.170;user=phone>
> Contact: <sip:100@172.22.0.199:5060>
> Content-Length: 0
>
------------------------------------------------------------------------------
>
>
>
>
>


--

Anton Yurchenko<phila@dg.net.ua>
Digital Generation