[asterisk-bugs] [Asterisk 0015395]: Dialling Fast on SIP (484) Does not match Dialplan
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Jun 25 07:16:24 CDT 2009
The following issue has been SUBMITTED.
======================================================================
https://issues.asterisk.org/view.php?id=15395
======================================================================
Reported By: leobrown
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 15395
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.6.0.6
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-06-25 07:16 CDT
Last Modified: 2009-06-25 07:16 CDT
======================================================================
Summary: Dialling Fast on SIP (484) Does not match Dialplan
Description:
I am using the asterisk 484 response on a SIP device to control dial
patterns. In this mode, every digit dialled is sent, and only when the
number is matched by Asterisk is the call connected.
However, I am finding that for a short pattern (00 in this case), if the
two digits are dialled faster than about 200ms then the dialplan pattern
for these two digits are not matched.
The dialplan is a very simple form:
00 => {
Answer();
Read(number);
}
The slow (correct) form is shown just below. The incorrect (fast) form is
shown in additional information.
I have attempted a work around using _00. and then adding ${EXTEN} to a
read variable, but I again get strange results when dialing fast.
<--- SIP read from UDP://87.81.167.157:5062 --->
INVITE sip:0 at trunk1.netfuse.org SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5062;branch=z9hG4bK442b133eade944e1;rport
From: "Leo Brown"
<sip:leo_kitchen at trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:0 at trunk1.netfuse.org>
Contact: <sip:leo_kitchen at 87.81.167.157:5062;transport=udp>
Supported: replaces, timer, path
Call-ID: 8c0a8e3ebbf4d665 at 10.10.1.4
CSeq: 61064 INVITE
User-Agent: Grandstream GXP2000 1.1.6.37
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 188
v=0
o=leo_kitchen 8000 8000 IN IP4 87.81.167.157
s=SIP Call
c=IN IP4 87.81.167.157
t=0 0
m=audio 5004 RTP/AVP 0 8
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
--- (13 headers 10 lines) ---
Sending to 87.81.167.157 : 5062 (NAT)
Using INVITE request as basis request - 8c0a8e3ebbf4d665 at 10.10.1.4
Found user 'leo_kitchen' for 'leo_kitchen'
<--- Reliably Transmitting (NAT) to 87.81.167.157:5062 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.10.1.4:5062;branch=z9hG4bK442b133eade944e1;received=87.81.167.157;rport=5062
From: "Leo Brown"
<sip:leo_kitchen at trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:0 at trunk1.netfuse.org>;tag=as4f02f000
Call-ID: 8c0a8e3ebbf4d665 at 10.10.1.4
CSeq: 61064 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="40b9b6a8"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '8c0a8e3ebbf4d665 at 10.10.1.4' in 32000
ms (Method: INVITE)
trunk1*CLI>
<--- SIP read from UDP://87.81.167.157:5062 --->
ACK sip:0 at trunk1.netfuse.org SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5062;branch=z9hG4bK442b133eade944e1;rport
From: "Leo Brown"
<sip:leo_kitchen at trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:0 at trunk1.netfuse.org>;tag=as4f02f000
Contact: <sip:leo_kitchen at 87.81.167.157:5062;transport=udp>
Supported: path
Call-ID: 8c0a8e3ebbf4d665 at 10.10.1.4
CSeq: 61064 ACK
User-Agent: Grandstream GXP2000 1.1.6.37
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
trunk1*CLI>
<--- SIP read from UDP://87.81.167.157:5062 --->
INVITE sip:0 at trunk1.netfuse.org SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5062;branch=z9hG4bKdb26531f412fd122;rport
From: "Leo Brown"
<sip:leo_kitchen at trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:0 at trunk1.netfuse.org>
Contact: <sip:leo_kitchen at 87.81.167.157:5062;transport=udp>
Supported: replaces, timer, path
Authorization: Digest username="leo_kitchen", realm="asterisk",
algorithm=MD5, uri="sip:0 at trunk1.netfuse.org", nonce="40b9b6a8",
response="bd0834f23cadeadae6874336898f00b9"
Call-ID: 8c0a8e3ebbf4d665 at 10.10.1.4
CSeq: 61065 INVITE
User-Agent: Grandstream GXP2000 1.1.6.37
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 188
v=0
o=leo_kitchen 8000 8001 IN IP4 87.81.167.157
s=SIP Call
c=IN IP4 87.81.167.157
t=0 0
m=audio 5004 RTP/AVP 0 8
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
--- (14 headers 10 lines) ---
Sending to 87.81.167.157 : 5062 (NAT)
Using INVITE request as basis request - 8c0a8e3ebbf4d665 at 10.10.1.4
Found user 'leo_kitchen' for 'leo_kitchen'
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port 87.81.167.157:5004
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 87.81.167.157:5004
Looking for 0 in acumen (domain trunk1.netfuse.org)
<--- Reliably Transmitting (NAT) to 87.81.167.157:5062 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP
10.10.1.4:5062;branch=z9hG4bKdb26531f412fd122;received=87.81.167.157;rport=5062
From: "Leo Brown"
<sip:leo_kitchen at trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:0 at trunk1.netfuse.org>;tag=as4f02f000
Call-ID: 8c0a8e3ebbf4d665 at 10.10.1.4
CSeq: 61065 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '8c0a8e3ebbf4d665 at 10.10.1.4' in 32000
ms (Method: INVITE)
trunk1*CLI>
<--- SIP read from UDP://87.81.167.157:5062 --->
ACK sip:0 at trunk1.netfuse.org SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5062;branch=z9hG4bKdb26531f412fd122;rport
From: "Leo Brown"
<sip:leo_kitchen at trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:0 at trunk1.netfuse.org>;tag=as4f02f000
Contact: <sip:leo_kitchen at 87.81.167.157:5062;transport=udp>
Supported: path
Authorization: Digest username="leo_kitchen", realm="asterisk",
algorithm=MD5, uri="sip:0 at trunk1.netfuse.org", nonce="40b9b6a8",
response="bd0834f23cadeadae6874336898f00b9"
Call-ID: 8c0a8e3ebbf4d665 at 10.10.1.4
CSeq: 61065 ACK
User-Agent: Grandstream GXP2000 1.1.6.37
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
trunk1*CLI>
<--- SIP read from UDP://87.81.167.157:5062 --->
INVITE sip:00 at trunk1.netfuse.org SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5062;branch=z9hG4bK597deb3c9b6c1bc3;rport
From: "Leo Brown"
<sip:leo_kitchen at trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:00 at trunk1.netfuse.org>;tag=as4f02f000
Contact: <sip:leo_kitchen at 87.81.167.157:5062;transport=udp>
Supported: replaces, timer, path
Authorization: Digest username="leo_kitchen", realm="asterisk",
algorithm=MD5, uri="sip:00 at trunk1.netfuse.org", nonce="40b9b6a8",
response="05f100a9753f43d2b51271c651c3474f"
Call-ID: 8c0a8e3ebbf4d665 at 10.10.1.4
CSeq: 61066 INVITE
User-Agent: Grandstream GXP2000 1.1.6.37
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 188
v=0
o=leo_kitchen 8000 8002 IN IP4 87.81.167.157
s=SIP Call
c=IN IP4 87.81.167.157
t=0 0
m=audio 5004 RTP/AVP 0 8
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
--- (14 headers 10 lines) ---
Sending to 87.81.167.157 : 5062 (NAT)
Using INVITE request as basis request - 8c0a8e3ebbf4d665 at 10.10.1.4
Found user 'leo_kitchen' for 'leo_kitchen'
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port 87.81.167.157:5004
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 87.81.167.157:5004
Looking for 00 in acumen (domain trunk1.netfuse.org)
list_route: hop: <sip:leo_kitchen at 87.81.167.157:5062;transport=udp>
<--- Transmitting (NAT) to 87.81.167.157:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.10.1.4:5062;branch=z9hG4bK597deb3c9b6c1bc3;received=87.81.167.157;rport=5062
From: "Leo Brown"
<sip:leo_kitchen at trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:00 at trunk1.netfuse.org>;tag=as4f02f000
Call-ID: 8c0a8e3ebbf4d665 at 10.10.1.4
CSeq: 61066 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:00 at 85.13.242.9>
Content-Length: 0
<------------>
-- Executing [00 at acumen:1] Answer("SIP/leo_kitchen-08315000", "") in
new stack
Audio is at 85.13.242.9 port 17842
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
<--- Reliably Transmitting (NAT) to 87.81.167.157:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.10.1.4:5062;branch=z9hG4bK597deb3c9b6c1bc3;received=87.81.167.157;rport=5062
From: "Leo Brown"
<sip:leo_kitchen at trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:00 at trunk1.netfuse.org>;tag=as4f02f000
Call-ID: 8c0a8e3ebbf4d665 at 10.10.1.4
CSeq: 61066 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:00 at 85.13.242.9>
Content-Type: application/sdp
Content-Length: 229
v=0
o=root 2112250526 2112250526 IN IP4 85.13.242.9
s=Asterisk PBX 1.6.0.9
c=IN IP4 85.13.242.9
t=0 0
m=audio 17842 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
trunk1*CLI>
<--- SIP read from UDP://87.81.167.157:5062 --->
ACK sip:00 at 85.13.242.9 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5062;branch=z9hG4bK719964f91a152b46;rport
From: "Leo Brown"
<sip:leo_kitchen at trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:00 at trunk1.netfuse.org>;tag=as4f02f000
Contact: <sip:leo_kitchen at 87.81.167.157:5062;transport=udp>
Supported: path
Authorization: Digest username="leo_kitchen", realm="asterisk",
algorithm=MD5, uri="sip:00 at trunk1.netfuse.org", nonce="40b9b6a8",
response="05f100a9753f43d2b51271c651c3474f"
Call-ID: 8c0a8e3ebbf4d665 at 10.10.1.4
CSeq: 61066 ACK
User-Agent: Grandstream GXP2000 1.1.6.37
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
-- Executing [00 at acumen:2] Read("SIP/leo_kitchen-08315000", "number")
in new stack
======================================================================
Issue History
Date Modified Username Field Change
======================================================================
2009-06-25 07:16 leobrown New Issue
2009-06-25 07:16 leobrown Asterisk Version => 1.6.0.6
2009-06-25 07:16 leobrown Regression => No
2009-06-25 07:16 leobrown SVN Branch (only for SVN checkouts, not tarball
releases) => N/A
======================================================================
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