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<DIV><FONT size=2><FONT size=3><FONT face=Arial size=2>Quick update on my
issues, Voicemail doesn't pickup also. It just drops the
line..</FONT></FONT></FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Thank you</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Chris Tuska</FONT></DIV>
<DIV><FONT size=2><FONT size=3><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>------------------------------</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT><BR>Hello All,<BR><BR>I have Cisco 7960's,
Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out
no issues but when someone calls in the phone rings I answer and the phone
disconnects the call. Call from my cell to my house I answer the cisco
phone it then disconnects the call at the same time on the cell I hear 4 beeps
and about 5 secs later the line on the cell drops, as anyone seen this?
<BR><BR>Thanks for the help..<BR><BR>Chris Tuska<BR><BR>***NOTE: Debug
Info first then Confs after...<BR><BR>linux01*CLI> sip show
peers<BR>Name/username
Host Dyn Nat
ACL Mask
Port Status
<BR>303/303
10.0.0.46
D 255.255.255.255
5060
Unmonitored<BR>203/203
10.0.0.46
D 255.255.255.255
5060 Unmonitored<BR>Sipmedia/970378
69.1.236.33
255.255.255.255 5060
Unmonitored<BR>linux01*CLI> <BR><BR>linux01*CLI> sip debug peer 203<BR>SIP
Debugging Enabled for IP: 10.0.0.46:5060<BR>linux01*CLI> sip debug peer
Sipmedia<BR>SIP Debugging Enabled for IP: 69.1.236.33:5060<BR>linux01*CLI>
<BR><BR>Sip read: <BR>INVITE sip:s@10.0.0.245:5060 SIP/2.0<BR>Record-Route:
<sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr><BR>Record-Route:
<sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr><BR>Via:
SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2<BR>Via: SIP/2.0/TCP
209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164<BR>From:
<sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637<BR>To:
<sip:+1Myphonenumber@69.1.236.33><BR>Call-ID: </FONT><A href=""><FONT
size=3>DEN0032050080410900407@209.244.63.17</FONT></A><BR><FONT size=3>CSeq: 1
INVITE<BR>Contact: <sip:209.247.16.5:5060;transport=tcp><BR>Max-Forwards:
68<BR>Content-Type: application/sdp<BR>Content-Length: 119<BR>Remote-Party-ID:
<sip:+1Mycellnumber@209.244.63.17>;party=calling;screen=yes;privacy=off<BR><BR>v=0<BR>o=-
1105159869 1105159870 IN IP4 209.247.23.201<BR>s=-<BR>c=IN IP4
209.247.23.201<BR>t=0 0<BR>m=audio 60062 RTP/AVP 0 18<BR><BR>14 headers, 6
lines<BR>Using latest request as basis request<BR>Sending to 69.1.236.33 : 5060
(non-NAT)<BR>Found RTP audio format 0<BR>Found RTP audio format 18<BR>Peer audio
RTP is at port 209.247.23.201:60062<BR>Capabilities: us - 0xe (gsm|ulaw|alaw),
peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4
(ulaw)<BR>Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing),
combined - 0x0 (nothing)<BR>Found peer 'Sipmedia'<BR>Looking for s in
from-Sipmedia<BR>list_route: hop:
<sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr><BR>list_route:
hop:
<sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr><BR>list_route:
hop: <sip:209.247.16.5:5060;transport=tcp><BR>Transmitting (no
NAT):<BR>SIP/2.0 100 Trying<BR>Via: SIP/2.0/UDP
69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2<BR>Via: SIP/2.0/TCP
209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164<BR>From:
<sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637<BR>To:
<sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0<BR>Call-ID: </FONT><A
href=""><FONT size=3>DEN0032050080410900407@209.244.63.17</FONT></A><BR><FONT
size=3>CSeq: 1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER<BR>Contact: <sip:s@10.0.0.245><BR>Content-Length:
0<BR><BR><BR> to 69.1.236.33:5060<BR>Transmitting (no NAT):<BR>SIP/2.0 180
Ringing<BR>Via: SIP/2.0/UDP
69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2<BR>Via: SIP/2.0/TCP
209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164<BR>From:
<sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637<BR>To:
<sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0<BR>Call-ID: </FONT><A
href=""><FONT size=3>DEN0032050080410900407@209.244.63.17</FONT></A><BR><FONT
size=3>CSeq: 1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER<BR>Contact: <sip:s@10.0.0.245><BR>Content-Length:
0<BR><BR><BR> to 69.1.236.33:5060<BR>We're at 10.0.0.245 port
11458<BR>Answering with preferred capability 0x2 (gsm)<BR>Answering with
preferred capability 0x4 (ulaw)<BR>Answering with preferred capability 0x8
(alaw)<BR>Reliably Transmitting (no NAT):<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2<BR>Via: SIP/2.0/TCP
209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164<BR>Record-Route:
<sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr><BR>Record-Route:
<sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr><BR>From:
<sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637<BR>To:
<sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0<BR>Call-ID: </FONT><A
href=""><FONT size=3>DEN0032050080410900407@209.244.63.17</FONT></A><BR><FONT
size=3>CSeq: 1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER<BR>Contact: <sip:s@10.0.0.245><BR>Content-Type:
application/sdp<BR>Content-Length: 201<BR><BR>v=0<BR>o=root 4696 4696 IN IP4
10.0.0.245<BR>s=session<BR>c=IN IP4 10.0.0.245<BR>t=0 0<BR>m=audio 11458 RTP/AVP
3 0 8<BR>a=rtpmap:3 GSM/8000<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:8
PCMA/8000<BR>a=silenceSupp:off - - - -<BR><BR> to
69.1.236.33:5060<BR>linux01*CLI> <BR><BR>Sip read: <BR>ACK
sip:s@10.0.0.245:5060 SIP/2.0<BR>Record-Route:
<sip:s@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr><BR>Record-Route:
<sip:s@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr><BR>Via:
SIP/2.0/UDP 69.1.236.33;branch=0;i=2<BR>Via: SIP/2.0/TCP
209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419168<BR>From:
<sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637<BR>To:
<sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0<BR>Call-ID: </FONT><A
href=""><FONT size=3>DEN0032050080410900407@209.244.63.17</FONT></A><BR><FONT
size=3>CSeq: 1 ACK<BR>Contact:
<sip:209.247.16.5:5060;transport=tcp><BR>Max-Forwards:
69<BR>Content-Length: 0<BR><BR><BR>12 headers, 0 lines<BR>linux01*CLI>
<BR><BR>Sip read: <BR>BYE sip:s@10.0.0.245:5060 SIP/2.0<BR>Record-Route:
<sip:s@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr><BR>Record-Route:
<sip:s@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr><BR>Via:
SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK0a8d.0eeb0ca4.0;i=2<BR>Via: SIP/2.0/TCP
209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419169<BR>From:
<sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637<BR>To:
<sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0<BR>Call-ID: </FONT><A
href=""><FONT size=3>DEN0032050080410900407@209.244.63.17</FONT></A><BR><FONT
size=3>CSeq: 2 BYE<BR>Contact:
<sip:209.247.16.5:5060;transport=tcp><BR>Max-Forwards:
67<BR>Content-Length: 0<BR><BR><BR>12 headers, 0 lines<BR>Sending to 69.1.236.33
: 5060 (non-NAT)<BR>Transmitting (no NAT):<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
69.1.236.33;branch=z9hG4bK0a8d.0eeb0ca4.0;i=2<BR>Via: SIP/2.0/TCP
209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419169<BR>Record-Route:
<sip:s@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr><BR>Record-Route:
<sip:s@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr><BR>From:
<sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637<BR>To:
<sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0<BR>Call-ID: </FONT><A
href=""><FONT size=3>DEN0032050080410900407@209.244.63.17</FONT></A><BR><FONT
size=3>CSeq: 2 BYE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER<BR>Contact: <sip:s@10.0.0.245><BR>Content-Length:
0<BR><BR><BR> to 69.1.236.33:5060<BR>Destroying call </FONT><A
href=""><FONT size=3>'DEN0032050080410900407@209.244.63.17'</FONT></A><BR><FONT
size=3>linux01*CLI> <BR><BR>Sip read: <BR>BYE sip:s@10.0.0.245:5060
SIP/2.0<BR>Record-Route:
<sip:s@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr><BR>Record-Route:
<sip:s@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr><BR>Via:
SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK0a8d.0eeb0ca4.0;i=2<BR>Via: SIP/2.0/TCP
209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419169<BR>From:
<sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637<BR>To:
<sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0<BR>Call-ID: </FONT><A
href=""><FONT size=3>DEN0032050080410900407@209.244.63.17</FONT></A><BR><FONT
size=3>CSeq: 2 BYE<BR>Contact:
<sip:209.247.16.5:5060;transport=tcp><BR>Max-Forwards:
67<BR>Content-Length: 0<BR><BR><BR>12 headers, 0 lines<BR>Sending to 69.1.236.33
: 5060 (non-NAT)<BR>Transmitting (no NAT):<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
69.1.236.33;branch=z9hG4bK0a8d.0eeb0ca4.0;i=2<BR>Via: SIP/2.0/TCP
209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419169<BR>Record-Route:
<sip:s@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr><BR>Record-Route:
<sip:s@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr><BR>From:
<sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637<BR>To:
<sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0<BR>Call-ID: </FONT><A
href=""><FONT size=3>DEN0032050080410900407@209.244.63.17</FONT></A><BR><FONT
size=3>CSeq: 2 BYE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER<BR>Contact: <BR>Content-Length: 0<BR><BR><BR> to
69.1.236.33:5060<BR>Destroying call </FONT><A href=""><FONT
size=3>'DEN0032050080410900407@209.244.63.17'</FONT></A><BR><FONT
size=3>linux01*CLI> sip no debug<BR>SIP Debugging
Disabled<BR><BR>linux01:/etc/asterisk # cat extensions.conf<BR>; Tuska
extensions.conf Dec 24,2004<BR>; Change to
Sipmedia<BR>;<BR>[general]<BR>;<BR>static=yes<BR>;<BR>writeprotect=yes<BR>;<BR><BR>;[globals]<BR><BR>;[bogon-calls]<BR>;<BR>;<BR>;
Take unknown callers that may have found<BR>; our system, and send them to a
re-order tone.<BR>; The string "_." matches any dialed sequence, so all<BR>;
calls will result in the Congestion tone application<BR>; being called. They'll
get bored and hang up eventually.<BR>;<BR>;<BR>;exten => _.,1,Congestion
<BR><BR>[default]<BR>;Extension 200 Cordless Phone<BR>exten =>
200,1,Dial(SIP/200,20)<BR>exten => 200,2,Voicemail(u200)<BR>exten =>
200,102,Voicemail(b200)<BR>exten => 200,103,Hangup<BR><BR>;Extension 203
Office Phone<BR>exten => 203,1,Dial(SIP/203,20)<BR>exten =>
203,2,Voicemail(u200)<BR>exten => 203,102,Voicemail(b200)<BR>exten =>
203,103,Hangup<BR><BR>;Extension 303 Office Phone<BR>exten =>
303,1,Dial(SIP/303,20)<BR>exten => 303,103,Hangup<BR><BR>; Voicemail
number<BR>exten =>
299,1,VoicemailMain(${CALLERIDNUM})<BR><BR>;sipmedia_outbound<BR>exten =>
_1NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@Sipmedia)<BR>exten =>
_1NXXNXXXXXX,4,Congestion()<BR>exten =>
_1NXXNXXXXXX,102,Busy()<BR><BR>;[conference] <BR>;exten => 300,1,AGI(callall)
<BR>;exten => 300,2,MeetMe(300,dtqp) ; press # to exit the conference
<BR>;exten => 300,3,MeetMeAdmin(300,K) ; kick all users out <BR>;exten =>
300,4,Hangup <BR>;exten => h,1,Hangup <BR>; <BR>;[add-to-conference]
<BR>;exten => start,1,MeetMe(300,dmqp) <BR>;exten => h,1,Hangup
<BR><BR><BR>[from-Sipmedia]<BR>exten =>
s,1,Dial(SIP/200&SIP/203,40,tr)<BR>exten => s,2,Voicemail(u200)<BR>exten
=> s,102,Voicemail(b200)<BR>exten =>
s,103,Hangup<BR>----end-----<BR><BR>linux01:/etc/asterisk # cat sip.conf<BR>;
Tuska extensions.conf Dec 24,2004<BR>; Change to Sipmedia<BR>;<BR>; SIP
Configuration for
Asterisk<BR>;<BR>[general]<BR>disallow=all<BR>allow=gsm<BR>allow=ulaw<BR>allow=alaw<BR>port=5060
; Port to bind
to<BR>context=default
; Default for incoming
calls<BR>bindaddr=10.0.0.245
; IP address to bind to (0.0.0.0 binds to
all)<BR>maxexpirey=180
; Maximum expiration for
registrations<BR>defaultexpirey=160
; Default expiration for
registrations<BR>canreinvite=no
; Allow clients to directly connect if set to yes. Set to no if behind
NAT.<BR>tos=reliability<BR>srvlookup=yes
; Enable DNS SRV lookups on outbound
calls<BR>videosupport=no
; Turn on support for SIP
video<BR>dtmfmode=inband
; DTMF inband need to be set here. If you are going to be using a<BR>;
nat=yes
; NAT settings <BR>register => #####:pass:#####@sip.sipmedia.com<BR><BR>; My
PSTN Service
provider<BR><BR>[Sipmedia]<BR>type=friend<BR>username=####<BR>fromuser=#####<BR>secret=password<BR>host=sip.sipmedia.com<BR>disallow=all<BR>allow=gsm<BR>allow=ulaw<BR>allow=alaw<BR>context=from-Sipmedia<BR>realm=sip1.xchangetele.com<BR>fromdomain=sip.sipmedia.com<BR>dtmfmode=inband<BR>canreinvite=no<BR>insecure=very<BR><BR>[200]<BR>type=friend<BR>username=200<BR>secret=pass<BR>callerid="Coreless
Phone"
<200><BR>mailbox=200<BR>host=dynamic<BR>;context=fromcisco<BR>;context=intern<BR>canreinvite=no<BR>dtmfmode=rfc2833<BR>disallow=all<BR>allow=ulaw<BR><BR>[203]<BR>type=friend<BR>username=203<BR>secret=pass<BR>callerid="Office
Phone"
<203><BR>;mailbox=203<BR>host=dynamic<BR>dtmfmode=rfc2833<BR>;context=fromcisco<BR>canreinvite=no<BR>disallow=all<BR>allow=ulaw<BR><BR>[303]<BR>type=friend<BR>username=303<BR>secret=pass<BR>callerid="Office
Phone"
<303><BR>host=dynamic<BR>dtmfmode=rfc2833<BR>canreinvite=no<BR>disallow=all<BR>allow=ulaw<BR>----end---<BR></DIV></FONT></FONT></BODY></HTML>