Hi, I started having problems recently when making calls to cell phones. I make the call, it rings, and then, instead of following the dial plan it hangs up. In the debug i see that im sending CANCEL, without any apparent reason. So I assume there must be a timeout. I don’t have that problem when phoning land lines, (where the call setup time is lower).
I don’t think i made any change in my asterisk box.
I would like to know if there is a way to make the timeout time longer, to test if that is actually the problem or not.
Im using asterisk 1.4.22.
Thanks!
Laura
Sip Debug:
-- Attempting call on sip/is/0825540216 for s@outbound:1 (Retry 1)
Audio is at 196.36.243.210 port 11918
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.3.14:5060:
INVITE sip:0825540216@192.168.3.14 SIP/2.0
Via: SIP/2.0/UDP 196.36.243.210:5060;branch=z9hG4bK7ed975ca;rport
From: “Grapevine test for IS” sip:0873514368@196.36.243.210;tag=as2a17094c
To: sip:0825540216@192.168.3.14
Contact: sip:0873514368@196.36.243.210
Call-ID: 73ee4e26518cd2d7171d47c9232523e9@196.36.243.210
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 03 Aug 2009 10:59:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 288
v=0
o=root 3145 3145 IN IP4 196.36.243.210
s=session
c=IN IP4 196.36.243.210
t=0 0
m=audio 11918 RTP/AVP 18 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
gvrhtst11*CLI>
<— SIP read from 192.168.3.14:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 196.36.243.210:5060;branch=z9hG4bK7ed975ca;rport=5060
From: “Grapevine test for IS” sip:0873514368@196.36.243.210;tag=as2a17094c
To: sip:0825540216@192.168.3.14
Call-ID: 73ee4e26518cd2d7171d47c9232523e9@196.36.243.210
CSeq: 102 INVITE
<------------->
— (6 headers 0 lines) —
Really destroying SIP dialog ‘24137bc800d525fb3d58d9e45c46f533@196.36.243.210’ Method: OPTIONS
gvrhtst11*CLI>
<— SIP read from 192.168.3.14:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 196.36.243.210:5060;branch=z9hG4bK7ed975ca;rport=5060
From: “Grapevine test for IS” sip:0873514368@196.36.243.210;tag=as2a17094c
To: sip:0825540216@192.168.3.14;tag=SD6vnc299-1912888113-1249289416833
Call-ID: 73ee4e26518cd2d7171d47c9232523e9@196.36.243.210
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Supported:
Contact: sip:0825540216@192.168.3.14:5060;transport=udp
Remote-Party-ID: sip:0825540216@196.35.130.3;user=phone;screen=yes;party=called;privacy=off;id-type=subscriber
Content-Length: 0
<------------->
— (11 headers 0 lines) —
gvrhtst11*CLI>
<— SIP read from 192.168.3.14:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 196.36.243.210:5060;branch=z9hG4bK7ed975ca;rport=5060
From: “Grapevine test for IS” sip:0873514368@196.36.243.210;tag=as2a17094c
To: sip:0825540216@192.168.3.14;tag=SD6vnc299-1912888113-1249289416833
Call-ID: 73ee4e26518cd2d7171d47c9232523e9@196.36.243.210
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Supported:
Contact: sip:0825540216@192.168.3.14:5060;transport=udp
Remote-Party-ID: sip:0825540216@196.35.130.3;user=phone;screen=yes;party=called;privacy=off;id-type=subscriber
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Scheduling destruction of SIP dialog ‘73ee4e26518cd2d7171d47c9232523e9@196.36.243.210’ in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.3.14:5060:
CANCEL sip:0825540216@192.168.3.14 SIP/2.0
Via: SIP/2.0/UDP 196.36.243.210:5060;branch=z9hG4bK7ed975ca;rport
From: “Grapevine test for IS” sip:0873514368@196.36.243.210;tag=as2a17094c
To: sip:0825540216@192.168.3.14
Call-ID: 73ee4e26518cd2d7171d47c9232523e9@196.36.243.210
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Scheduling destruction of SIP dialog ‘73ee4e26518cd2d7171d47c9232523e9@196.36.243.210’ in 6400 ms (Method: INVITE)
[Aug 3 13:00:12] NOTICE[13713]: pbx_spool.c:355 attempt_thread: Call failed to go through, reason (3) Remote end Ringing
gvrhtst11*CLI>
<— SIP read from 192.168.3.14:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 196.36.243.210:5060;branch=z9hG4bK7ed975ca;rport=5060
From: “Grapevine test for IS” sip:0873514368@196.36.243.210;tag=as2a17094c
To: sip:0825540216@192.168.3.14;tag=SD6vnc299-1912888113-1249289416833
Call-ID: 73ee4e26518cd2d7171d47c9232523e9@196.36.243.210
CSeq: 102 CANCEL