Chris Tuska
2005-Jan-09 10:38 UTC
[Asterisk-Users] RE: Inbound calls getting disconnected when I answer the phone, using 'SIP'
Quick update on my issues, Voicemail doesn't pickup also. It just drops the
line..
Thank you
Chris Tuska
------------------------------
Hello All,
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?
Thanks for the help..
Chris Tuska
***NOTE: Debug Info first then Confs after...
linux01*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
303/303 10.0.0.46 D 255.255.255.255 5060
Unmonitored
203/203 10.0.0.46 D 255.255.255.255 5060
Unmonitored
Sipmedia/970378 69.1.236.33 255.255.255.255 5060
Unmonitored
linux01*CLI>
linux01*CLI> sip debug peer 203
SIP Debugging Enabled for IP: 10.0.0.46:5060
linux01*CLI> sip debug peer Sipmedia
SIP Debugging Enabled for IP: 69.1.236.33:5060
linux01*CLI>
Sip read:
INVITE sip:s@10.0.0.245:5060 SIP/2.0
Record-Route:
<sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
Record-Route:
<sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2
Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164
From: <sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637
To: <sip:+1Myphonenumber@69.1.236.33>
Call-ID: DEN0032050080410900407@209.244.63.17
CSeq: 1 INVITE
Contact: <sip:209.247.16.5:5060;transport=tcp>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 119
Remote-Party-ID:
<sip:+1Mycellnumber@209.244.63.17>;party=calling;screen=yes;privacy=off
v=0
o=- 1105159869 1105159870 IN IP4 209.247.23.201
s=-
c=IN IP4 209.247.23.201
t=0 0
m=audio 60062 RTP/AVP 0 18
14 headers, 6 lines
Using latest request as basis request
Sending to 69.1.236.33 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 209.247.23.201:60062
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x104 (ulaw|g729)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0
(nothing)
Found peer 'Sipmedia'
Looking for s in from-Sipmedia
list_route: hop:
<sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
list_route: hop:
<sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
list_route: hop: <sip:209.247.16.5:5060;transport=tcp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2
Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164
From: <sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637
To: <sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0
Call-ID: DEN0032050080410900407@209.244.63.17
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:s@10.0.0.245>
Content-Length: 0
to 69.1.236.33:5060
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2
Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164
From: <sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637
To: <sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0
Call-ID: DEN0032050080410900407@209.244.63.17
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:s@10.0.0.245>
Content-Length: 0
to 69.1.236.33:5060
We're at 10.0.0.245 port 11458
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2
Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164
Record-Route:
<sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
Record-Route:
<sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
From: <sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637
To: <sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0
Call-ID: DEN0032050080410900407@209.244.63.17
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:s@10.0.0.245>
Content-Type: application/sdp
Content-Length: 201
v=0
o=root 4696 4696 IN IP4 10.0.0.245
s=session
c=IN IP4 10.0.0.245
t=0 0
m=audio 11458 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
to 69.1.236.33:5060
linux01*CLI>
Sip read:
ACK sip:s@10.0.0.245:5060 SIP/2.0
Record-Route: <sip:s@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
Record-Route:
<sip:s@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
Via: SIP/2.0/UDP 69.1.236.33;branch=0;i=2
Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419168
From: <sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637
To: <sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0
Call-ID: DEN0032050080410900407@209.244.63.17
CSeq: 1 ACK
Contact: <sip:209.247.16.5:5060;transport=tcp>
Max-Forwards: 69
Content-Length: 0
12 headers, 0 lines
linux01*CLI>
Sip read:
BYE sip:s@10.0.0.245:5060 SIP/2.0
Record-Route: <sip:s@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
Record-Route:
<sip:s@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK0a8d.0eeb0ca4.0;i=2
Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419169
From: <sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637
To: <sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0
Call-ID: DEN0032050080410900407@209.244.63.17
CSeq: 2 BYE
Contact: <sip:209.247.16.5:5060;transport=tcp>
Max-Forwards: 67
Content-Length: 0
12 headers, 0 lines
Sending to 69.1.236.33 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK0a8d.0eeb0ca4.0;i=2
Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419169
Record-Route: <sip:s@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
Record-Route:
<sip:s@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
From: <sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637
To: <sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0
Call-ID: DEN0032050080410900407@209.244.63.17
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:s@10.0.0.245>
Content-Length: 0
to 69.1.236.33:5060
Destroying call 'DEN0032050080410900407@209.244.63.17'
linux01*CLI>
Sip read:
BYE sip:s@10.0.0.245:5060 SIP/2.0
Record-Route: <sip:s@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
Record-Route:
<sip:s@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK0a8d.0eeb0ca4.0;i=2
Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419169
From: <sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637
To: <sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0
Call-ID: DEN0032050080410900407@209.244.63.17
CSeq: 2 BYE
Contact: <sip:209.247.16.5:5060;transport=tcp>
Max-Forwards: 67
Content-Length: 0
12 headers, 0 lines
Sending to 69.1.236.33 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK0a8d.0eeb0ca4.0;i=2
Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419169
Record-Route: <sip:s@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
Record-Route:
<sip:s@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
From: <sip:+1Mycellnumber@209.247.16.5>;tag=VPSF50603522629637
To: <sip:+1Myphonenumber@69.1.236.33>;tag=as6e603ce0
Call-ID: DEN0032050080410900407@209.244.63.17
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 69.1.236.33:5060
Destroying call 'DEN0032050080410900407@209.244.63.17'
linux01*CLI> sip no debug
SIP Debugging Disabled
linux01:/etc/asterisk # cat extensions.conf
; Tuska extensions.conf Dec 24,2004
; Change to Sipmedia
;
[general]
;
static=yes
;
writeprotect=yes
;
;[globals]
;[bogon-calls]
;
;
; Take unknown callers that may have found
; our system, and send them to a re-order tone.
; The string "_." matches any dialed sequence, so all
; calls will result in the Congestion tone application
; being called. They'll get bored and hang up eventually.
;
;
;exten => _.,1,Congestion
[default]
;Extension 200 Cordless Phone
exten => 200,1,Dial(SIP/200,20)
exten => 200,2,Voicemail(u200)
exten => 200,102,Voicemail(b200)
exten => 200,103,Hangup
;Extension 203 Office Phone
exten => 203,1,Dial(SIP/203,20)
exten => 203,2,Voicemail(u200)
exten => 203,102,Voicemail(b200)
exten => 203,103,Hangup
;Extension 303 Office Phone
exten => 303,1,Dial(SIP/303,20)
exten => 303,103,Hangup
; Voicemail number
exten => 299,1,VoicemailMain(${CALLERIDNUM})
;sipmedia_outbound
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@Sipmedia)
exten => _1NXXNXXXXXX,4,Congestion()
exten => _1NXXNXXXXXX,102,Busy()
;[conference]
;exten => 300,1,AGI(callall)
;exten => 300,2,MeetMe(300,dtqp) ; press # to exit the conference
;exten => 300,3,MeetMeAdmin(300,K) ; kick all users out
;exten => 300,4,Hangup
;exten => h,1,Hangup
;
;[add-to-conference]
;exten => start,1,MeetMe(300,dmqp)
;exten => h,1,Hangup
[from-Sipmedia]
exten => s,1,Dial(SIP/200&SIP/203,40,tr)
exten => s,2,Voicemail(u200)
exten => s,102,Voicemail(b200)
exten => s,103,Hangup
----end-----
linux01:/etc/asterisk # cat sip.conf
; Tuska extensions.conf Dec 24,2004
; Change to Sipmedia
;
; SIP Configuration for Asterisk
;
[general]
disallow=all
allow=gsm
allow=ulaw
allow=alaw
port=5060 ; Port to bind to
context=default ; Default for incoming calls
bindaddr=10.0.0.245 ; IP address to bind to (0.0.0.0 binds to all)
maxexpirey=180 ; Maximum expiration for registrations
defaultexpirey=160 ; Default expiration for registrations
canreinvite=no ; Allow clients to directly connect if set to
yes. Set to no if behind NAT.
tos=reliability
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
videosupport=no ; Turn on support for SIP video
dtmfmode=inband ; DTMF inband need to be set here. If you are
going to be using a
; nat=yes ; NAT settings
register => #####:pass:#####@sip.sipmedia.com
; My PSTN Service provider
[Sipmedia]
type=friend
username=####
fromuser=#####
secret=password
host=sip.sipmedia.com
disallow=all
allow=gsm
allow=ulaw
allow=alaw
context=from-Sipmedia
realm=sip1.xchangetele.com
fromdomain=sip.sipmedia.com
dtmfmode=inband
canreinvite=no
insecure=very
[200]
type=friend
username=200
secret=pass
callerid="Coreless Phone" <200>
mailbox=200
host=dynamic
;context=fromcisco
;context=intern
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=ulaw
[203]
type=friend
username=203
secret=pass
callerid="Office Phone" <203>
;mailbox=203
host=dynamic
dtmfmode=rfc2833
;context=fromcisco
canreinvite=no
disallow=all
allow=ulaw
[303]
type=friend
username=303
secret=pass
callerid="Office Phone" <303>
host=dynamic
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
----end---
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