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@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
Hi, What's the status of the Gnophone? www.gnophone.com has not been updated since April 3, 2002... regards, -- Jukka Tainio | Kase ry. http://www.kase.fi | tel: 06-8887222
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