[asterisk-bugs] [Asterisk-GUI 0010151]: AsteriskNOW DID into wrong context
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Aug 28 16:18:35 CDT 2007
The following issue has been REOPENED.
======================================================================
http://bugs.digium.com/view.php?id=10151
======================================================================
Reported By: dmgeurts
Assigned To: pari
======================================================================
Project: Asterisk-GUI
Issue ID: 10151
Category: General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.4.5
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!): 1127
Disclaimer on File?: No
Request Review:
======================================================================
Date Submitted: 07-07-2007 10:16 CDT
Last Modified: 08-28-2007 16:18 CDT
======================================================================
Summary: AsteriskNOW DID into wrong context
Description:
Two accounts from same ITSP, each with a different DID number. When both
accounts are active and registered, only one picks up the call. This
reproducible and consisted. It is always the same trunk/context that the
incoming call is put into.
Disabling either account will result in correct operation, only reachable
on the DID which is registered not the other.
I have no workaround as I can't manage to catch the call using the called
number. Debugging shows 's' and in the wrong context.
======================================================================
----------------------------------------------------------------------
dmgeurts - 08-28-07 16:18
----------------------------------------------------------------------
Now at svn r1464. And with time to run a debug. 1st one is of a call coming
in from trunk_3 but being picked up by trunk_2. 2nd is a call coming in
from trunk_2, being picked up by trunk_2
trunk_3/31105117619 81.23.228.150 5060 OK (3 ms)
trunk_2/31107142242 81.23.228.150 5060 OK (3 ms)
My temporary fix is adding a line to exentions.conf:
[DID_trunk_2]
include = default
exten = _X.,1,Goto(default|102|1)
exten = s,1,Goto(default|102|1)
[DID_trunk_3]
include = default
exten = _31107142242,1,Goto(ringroups-custom-3,s,1) <<<<< goes to 102 and
200
exten = _X.,1,Goto(default|200|1)
exten = s,1,Goto(default|200|1)
<debug-http://bugs.digium.com/view.php?id=1>
sip*CLI>
<--- SIP read from 81.23.228.150:5060 --->
INVITE sip:31107142242 at 217.195.248.252 SIP/2.0
Record-Route: <sip:81.23.228.150;lr=on;ftag=as0db7a7b0;fcd=yes>
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK2916.d16f7711.0
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK486155f1;rport=5060
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as0db7a7b0
To: <sip:31107142242 at budgetphone.nl>
Contact: <sip:0651455735 at 80.252.84.179>
Call-ID: 647f00f652dac241778170f37084f69d at gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Tue, 28 Aug 2007 21:33:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 391
v=0
o=root 17847 17847 IN IP4 80.252.84.179
s=session
c=IN IP4 81.23.228.147
t=0 0
m=audio 63684 RTP/AVP 3 18 5 8 0 97 110 101
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:5 DVI4/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<------------->
--- (15 headers 17 lines) ---
Sending to 81.23.228.150 : 5060 (no NAT)
Using INVITE request as basis request -
647f00f652dac241778170f37084f69d at gw02-mci.budgetphone.nl
Found peer 'trunk_3'
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 101
Peer audio RTP is at port 81.23.228.147:63684
Found description format GSM for ID 3
Found description format G729 for ID 18
Found description format DVI4 for ID 5
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format iLBC for ID 97
Found description format speex for ID 110
Found description format telephone-event for ID 101
Capabilities: us - 0x3f1fff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264),
peer - audio=0x72e (gsm|ulaw|alaw|adpcm|g729|speex|ilbc)/video=0x0
(nothing), combined - 0x72e (gsm|ulaw|alaw|adpcm|g729|speex|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 81.23.228.147:63684
Peer video RTP is at port 81.23.228.147:5128
Looking for 31107142242 in DID_trunk_3 (domain 217.195.248.252)
list_route: hop: <sip:81.23.228.150;lr=on;ftag=as0db7a7b0;fcd=yes>
<--- Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
81.23.228.150;branch=z9hG4bK2916.d16f7711.0;received=81.23.228.150
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK486155f1;rport=5060
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as0db7a7b0
To: <sip:31107142242 at budgetphone.nl>
Call-ID: 647f00f652dac241778170f37084f69d at gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:31107142242 at 217.195.248.252>
Content-Length: 0
<------------>
-- Executing [31107142242 at DID_trunk_3:1]
Goto("SIP/31105117619-08aa1c70", "ringroups-custom-3|s|1") in new stack
-- Goto (ringroups-custom-3,s,1)
-- Executing [s at ringroups-custom-3:1] NoOp("SIP/31105117619-08aa1c70",
"RINGGROUP") in new stack
-- Executing [s at ringroups-custom-3:2] Dial("SIP/31105117619-08aa1c70",
"SIP/102&SIP/200|20") in new stack
-- Called 102
-- Called 200
-- SIP/200-08aaef10 is ringing
<--- Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
81.23.228.150;branch=z9hG4bK2916.d16f7711.0;received=81.23.228.150
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK486155f1;rport=5060
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as0db7a7b0
To: <sip:31107142242 at budgetphone.nl>;tag=as2dffc194
Call-ID: 647f00f652dac241778170f37084f69d at gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:31107142242 at 217.195.248.252>
Content-Length: 0
<------------>
-- SIP/102-08aa85c0 is ringing
sip*CLI>
<--- SIP read from 81.23.228.150:5060 --->
CANCEL sip:31107142242 at 217.195.248.252 SIP/2.0
Record-Route: <sip:81.23.228.150;lr=on;ftag=as0db7a7b0>
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK2916.d16f7711.0
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK486155f1;rport=5060
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as0db7a7b0
To: <sip:31107142242 at budgetphone.nl>
Contact: <sip:0651455735 at 80.252.84.179>
Call-ID: 647f00f652dac241778170f37084f69d at gw02-mci.budgetphone.nl
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 81.23.228.150 : 5060 (no NAT)
<--- Reliably Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
81.23.228.150;branch=z9hG4bK2916.d16f7711.0;received=81.23.228.150
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK486155f1;rport=5060
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as0db7a7b0
To: <sip:31107142242 at budgetphone.nl>;tag=as2dffc194
Call-ID: 647f00f652dac241778170f37084f69d at gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
81.23.228.150;branch=z9hG4bK2916.d16f7711.0;received=81.23.228.150
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK486155f1;rport=5060
Record-Route: <sip:81.23.228.150;lr=on;ftag=as0db7a7b0>
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as0db7a7b0
To: <sip:31107142242 at budgetphone.nl>;tag=as2dffc194
Call-ID: 647f00f652dac241778170f37084f69d at gw02-mci.budgetphone.nl
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:31107142242 at 217.195.248.252>
Content-Length: 0
<------------>
== Spawn extension (ringroups-custom-3, s, 2) exited non-zero on
'SIP/31105117619-08aa1c70'
sip*CLI>
<--- SIP read from 81.23.228.150:5060 --->
ACK sip:31107142242 at 217.195.248.252 SIP/2.0
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK2916.d16f7711.0
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as0db7a7b0
Call-ID: 647f00f652dac241778170f37084f69d at gw02-mci.budgetphone.nl
To: <sip:31107142242 at budgetphone.nl>;tag=as2dffc194
CSeq: 102 ACK
User-Agent: OpenSer (1.1.0-tls (i386/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog
'647f00f652dac241778170f37084f69d at gw02-mci.budgetphone.nl' Method: ACK
</debug-http://bugs.digium.com/view.php?id=1>
<debug-http://bugs.digium.com/view.php?id=2>
sip*CLI>
<--- SIP read from 81.23.228.150:5060 --->
INVITE sip:31105117619 at 217.195.248.252 SIP/2.0
Record-Route: <sip:81.23.228.150;lr=on;ftag=as40a7138d;fcd=yes>
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKf85c.8f168463.0
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK10b4824a;rport=5060
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as40a7138d
To: <sip:31105117619 at budgetphone.nl>
Contact: <sip:0651455735 at 80.252.84.179>
Call-ID: 17a433ea4f80abd750bb027e34ad7bd5 at gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Tue, 28 Aug 2007 21:34:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 391
v=0
o=root 17847 17847 IN IP4 80.252.84.179
s=session
c=IN IP4 83.149.75.105
t=0 0
m=audio 61642 RTP/AVP 3 18 5 8 0 97 110 101
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:5 DVI4/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<------------->
--- (15 headers 17 lines) ---
Sending to 81.23.228.150 : 5060 (no NAT)
Using INVITE request as basis request -
17a433ea4f80abd750bb027e34ad7bd5 at gw02-mci.budgetphone.nl
Found peer 'trunk_3'
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 101
Peer audio RTP is at port 83.149.75.105:61642
Found description format GSM for ID 3
Found description format G729 for ID 18
Found description format DVI4 for ID 5
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format iLBC for ID 97
Found description format speex for ID 110
Found description format telephone-event for ID 101
Capabilities: us - 0x3f1fff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264),
peer - audio=0x72e (gsm|ulaw|alaw|adpcm|g729|speex|ilbc)/video=0x0
(nothing), combined - 0x72e (gsm|ulaw|alaw|adpcm|g729|speex|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 83.149.75.105:61642
Peer video RTP is at port 83.149.75.105:5128
Looking for 31105117619 in DID_trunk_3 (domain 217.195.248.252)
list_route: hop: <sip:81.23.228.150;lr=on;ftag=as40a7138d;fcd=yes>
<--- Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
81.23.228.150;branch=z9hG4bKf85c.8f168463.0;received=81.23.228.150
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK10b4824a;rport=5060
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as40a7138d
To: <sip:31105117619 at budgetphone.nl>
Call-ID: 17a433ea4f80abd750bb027e34ad7bd5 at gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:31105117619 at 217.195.248.252>
Content-Length: 0
<------------>
-- Executing [31105117619 at DID_trunk_3:1]
Goto("SIP/31105117619-08a9b380", "default|200|1") in new stack
-- Goto (default,200,1)
-- Executing [200 at default:1] Macro("SIP/31105117619-08a9b380",
"stdexten|200|SIP/200") in new stack
-- Executing [s at macro-stdexten:1] Dial("SIP/31105117619-08a9b380",
"SIP/200|20") in new stack
-- Called 200
-- SIP/200-08aa2fc0 is ringing
<--- Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
81.23.228.150;branch=z9hG4bKf85c.8f168463.0;received=81.23.228.150
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK10b4824a;rport=5060
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as40a7138d
To: <sip:31105117619 at budgetphone.nl>;tag=as0bcd0cce
Call-ID: 17a433ea4f80abd750bb027e34ad7bd5 at gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:31105117619 at 217.195.248.252>
Content-Length: 0
<------------>
-- SIP/200-08aa2fc0 is ringing
sip*CLI>
<--- SIP read from 81.23.228.150:5060 --->
CANCEL sip:31105117619 at 217.195.248.252 SIP/2.0
Record-Route: <sip:81.23.228.150;lr=on;ftag=as40a7138d>
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKf85c.8f168463.0
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK10b4824a;rport=5060
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as40a7138d
To: <sip:31105117619 at budgetphone.nl>
Contact: <sip:0651455735 at 80.252.84.179>
Call-ID: 17a433ea4f80abd750bb027e34ad7bd5 at gw02-mci.budgetphone.nl
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 81.23.228.150 : 5060 (no NAT)
<--- Reliably Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
81.23.228.150;branch=z9hG4bKf85c.8f168463.0;received=81.23.228.150
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK10b4824a;rport=5060
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as40a7138d
To: <sip:31105117619 at budgetphone.nl>;tag=as0bcd0cce
Call-ID: 17a433ea4f80abd750bb027e34ad7bd5 at gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
81.23.228.150;branch=z9hG4bKf85c.8f168463.0;received=81.23.228.150
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK10b4824a;rport=5060
Record-Route: <sip:81.23.228.150;lr=on;ftag=as40a7138d>
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as40a7138d
To: <sip:31105117619 at budgetphone.nl>;tag=as0bcd0cce
Call-ID: 17a433ea4f80abd750bb027e34ad7bd5 at gw02-mci.budgetphone.nl
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:31105117619 at 217.195.248.252>
Content-Length: 0
<------------>
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/31105117619-08a9b380' in macro 'stdexten'
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/31105117619-08a9b380'
sip*CLI>
<--- SIP read from 81.23.228.150:5060 --->
ACK sip:31105117619 at 217.195.248.252 SIP/2.0
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKf85c.8f168463.0
From: "0651455735"
<sip:0651455735 at gw02-mci.budgetphone.nl>;tag=as40a7138d
Call-ID: 17a433ea4f80abd750bb027e34ad7bd5 at gw02-mci.budgetphone.nl
To: <sip:31105117619 at budgetphone.nl>;tag=as0bcd0cce
CSeq: 102 ACK
User-Agent: OpenSer (1.1.0-tls (i386/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog
'17a433ea4f80abd750bb027e34ad7bd5 at gw02-mci.budgetphone.nl' Method: ACK
</debug-http://bugs.digium.com/view.php?id=2>
Issue History
Date Modified Username Field Change
======================================================================
08-28-07 16:18 dmgeurts Status closed => feedback
08-28-07 16:18 dmgeurts Resolution unable to reproduce =>
reopened
08-28-07 16:18 dmgeurts Note Added: 0069585
======================================================================
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