Chris Tuska
2005-Jan-09 00:00 UTC
[Asterisk-Users] Inbound calls getting disconnected when I answer the phone, using 'SIP'.
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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Vassilis Konstantinou
2005-Jan-09 02:14 UTC
[Asterisk-Users] X100P random hangups - Please help with suggestions
This one is driving me crazy. So any suggestions will be very welcome. My setup: Suse Linux 9.0 (Pentium 4, 1GB) Asterisk: current stable (1.0.3?) - tried the head CVS before Christmas but did not fix it 2 X100P clones - one for a UK BT line, one connected to an ATA186 configured for a UK BT Broabband-Voice service (MGCP) 1 ATA186 (SIP) connected to two dect internal phones (configured as extensions 5000-5001) The problem: Both of the X100Ps seem to randomly hang-up both incoming and outgoing calls. There is no fixed dureation but it always happens. Sometimes as soon as a call is answered and sometimes at any point up to 10-15 minutes. All calls through my true VOIP lines (I use sipcall in UK and fwd) are fine and never disconnect during the call. The X100Ps seem to detect the "real" hangup properly (of course). Things I have tried: 1) The latest CVS (up to early December). No change 2) The current stable. No change 3) Playing with the rxgain in the zapata.conf file (no change) 4) Using the Loopstart instead of Kewlstart. No false hangups here BUT as expected lots of line noise. Is this a good clue to what is happening? Are there any parameters I can tweak to make the Kewlstart driver a bit more reliable? Please help. This is driving me (and the people using the system crazy). Vassilis My current zapata.conf is attached below: ======================= ;; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; group => 1 language=en ; ; Default context ; context=incoming switchtype=national usedistinctiveringdetection=no useincomingcalleridonzaptransfer=yes rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=no callwaitingcallerid=no threewaycalling=no transfer=yes cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=0.0 txgain=1.5 ; ; Specify whether the channel should be answered immediately or ; if the simple switch should provide dialtone, read digits, etc. ; immediate=no ;musiconhold=default callprogress=no progzone=uk ; usecallerid = yes ; we want Caller*ID support cidsignalling = v23 ; UK (BT) Caller*ID uses the V.23 std cidstart = history ; use the Zaptel history (X100P) busydetect=no signalling=fxs_ks channel => 1 ;BT Broadband Voice - Uses US ID and busysignal on Hangup busydetect=yes busycount=6 cidstart = ring ; ring starts Caller*ID cidsignalling = bell ; Cid US signalling=fxs_ks channel => 2
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