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Summary:ASTERISK-14373: Dialling Fast on SIP (484) Does not match Dialplan
Reporter:Leo Brown (netfuse)Labels:
Date Opened:2009-06-25 07:16:24Date Closed:2011-06-07 14:08:15
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
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@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@trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:0@trunk1.netfuse.org>
Contact: <sip:leo_kitchen@87.81.167.157:5062;transport=udp>
Supported: replaces, timer, path
Call-ID: 8c0a8e3ebbf4d665@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@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@trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:0@trunk1.netfuse.org>;tag=as4f02f000
Call-ID: 8c0a8e3ebbf4d665@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@10.10.1.4' in 32000 ms (Method: INVITE)
trunk1*CLI>
<--- SIP read from UDP://87.81.167.157:5062 --->
ACK sip:0@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@trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:0@trunk1.netfuse.org>;tag=as4f02f000
Contact: <sip:leo_kitchen@87.81.167.157:5062;transport=udp>
Supported: path
Call-ID: 8c0a8e3ebbf4d665@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@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@trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:0@trunk1.netfuse.org>
Contact: <sip:leo_kitchen@87.81.167.157:5062;transport=udp>
Supported: replaces, timer, path
Authorization: Digest username="leo_kitchen", realm="asterisk", algorithm=MD5, uri="sip:0@trunk1.netfuse.org", nonce="40b9b6a8", response="bd0834f23cadeadae6874336898f00b9"
Call-ID: 8c0a8e3ebbf4d665@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@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@trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:0@trunk1.netfuse.org>;tag=as4f02f000
Call-ID: 8c0a8e3ebbf4d665@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@10.10.1.4' in 32000 ms (Method: INVITE)
trunk1*CLI>
<--- SIP read from UDP://87.81.167.157:5062 --->
ACK sip:0@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@trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:0@trunk1.netfuse.org>;tag=as4f02f000
Contact: <sip:leo_kitchen@87.81.167.157:5062;transport=udp>
Supported: path
Authorization: Digest username="leo_kitchen", realm="asterisk", algorithm=MD5, uri="sip:0@trunk1.netfuse.org", nonce="40b9b6a8", response="bd0834f23cadeadae6874336898f00b9"
Call-ID: 8c0a8e3ebbf4d665@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@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@trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:00@trunk1.netfuse.org>;tag=as4f02f000
Contact: <sip:leo_kitchen@87.81.167.157:5062;transport=udp>
Supported: replaces, timer, path
Authorization: Digest username="leo_kitchen", realm="asterisk", algorithm=MD5, uri="sip:00@trunk1.netfuse.org", nonce="40b9b6a8", response="05f100a9753f43d2b51271c651c3474f"
Call-ID: 8c0a8e3ebbf4d665@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@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@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@trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:00@trunk1.netfuse.org>;tag=as4f02f000
Call-ID: 8c0a8e3ebbf4d665@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@85.13.242.9>
Content-Length: 0


<------------>
   -- Executing [00@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@trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:00@trunk1.netfuse.org>;tag=as4f02f000
Call-ID: 8c0a8e3ebbf4d665@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@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@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@trunk1.netfuse.org>;tag=25621cf32c2e44b8
To: <sip:00@trunk1.netfuse.org>;tag=as4f02f000
Contact: <sip:leo_kitchen@87.81.167.157:5062;transport=udp>
Supported: path
Authorization: Digest username="leo_kitchen", realm="asterisk", algorithm=MD5, uri="sip:00@trunk1.netfuse.org", nonce="40b9b6a8", response="05f100a9753f43d2b51271c651c3474f"
Call-ID: 8c0a8e3ebbf4d665@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@acumen:2] Read("SIP/leo_kitchen-08315000", "number") in new stack

****** ADDITIONAL INFORMATION ******

<--- SIP read from UDP://87.81.167.157:5062 --->
INVITE sip:0@trunk1.netfuse.org SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5062;branch=z9hG4bK2f0ac60d61d561d5;rport
From: "Leo Brown" <sip:leo_kitchen@trunk1.netfuse.org>;tag=b3fc348eb47a6ce3
To: <sip:0@trunk1.netfuse.org>
Contact: <sip:leo_kitchen@87.81.167.157:5062;transport=udp>
Supported: replaces, timer, path
Call-ID: e595ea6544af755b@10.10.1.4
CSeq: 57404 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 - e595ea6544af755b@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=z9hG4bK2f0ac60d61d561d5;received=87.81.167.157;rport=5062
From: "Leo Brown" <sip:leo_kitchen@trunk1.netfuse.org>;tag=b3fc348eb47a6ce3
To: <sip:0@trunk1.netfuse.org>;tag=as60dfe957
Call-ID: e595ea6544af755b@10.10.1.4
CSeq: 57404 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="077112db"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'e595ea6544af755b@10.10.1.4' in 32000 ms (Method: INVITE)
trunk1*CLI>
<--- SIP read from UDP://87.81.167.157:5062 --->
ACK sip:0@trunk1.netfuse.org SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5062;branch=z9hG4bK2f0ac60d61d561d5;rport
From: "Leo Brown" <sip:leo_kitchen@trunk1.netfuse.org>;tag=b3fc348eb47a6ce3
To: <sip:0@trunk1.netfuse.org>;tag=as60dfe957
Contact: <sip:leo_kitchen@87.81.167.157:5062;transport=udp>
Supported: path
Call-ID: e595ea6544af755b@10.10.1.4
CSeq: 57404 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@trunk1.netfuse.org SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5062;branch=z9hG4bK94fe5bba64e1d271;rport
From: "Leo Brown" <sip:leo_kitchen@trunk1.netfuse.org>;tag=b3fc348eb47a6ce3
To: <sip:0@trunk1.netfuse.org>
Contact: <sip:leo_kitchen@87.81.167.157:5062;transport=udp>
Supported: replaces, timer, path
Authorization: Digest username="leo_kitchen", realm="asterisk", algorithm=MD5, uri="sip:0@trunk1.netfuse.org", nonce="077112db", response="21fcde9e9df1f109040ddeba1c137ad8"
Call-ID: e595ea6544af755b@10.10.1.4
CSeq: 57405 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 - e595ea6544af755b@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=z9hG4bK94fe5bba64e1d271;received=87.81.167.157;rport=5062
From: "Leo Brown" <sip:leo_kitchen@trunk1.netfuse.org>;tag=b3fc348eb47a6ce3
To: <sip:0@trunk1.netfuse.org>;tag=as60dfe957
Call-ID: e595ea6544af755b@10.10.1.4
CSeq: 57405 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 'e595ea6544af755b@10.10.1.4' in 32000 ms (Method: INVITE)
trunk1*CLI>
<--- SIP read from UDP://87.81.167.157:5062 --->
INVITE sip:00@trunk1.netfuse.org SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5062;branch=z9hG4bK94fe5bba64e1d271;rport
From: "Leo Brown" <sip:leo_kitchen@trunk1.netfuse.org>;tag=b3fc348eb47a6ce3
To: <sip:00@trunk1.netfuse.org>
Contact: <sip:leo_kitchen@87.81.167.157:5062;transport=udp>
Supported: replaces, timer, path
Authorization: Digest username="leo_kitchen", realm="asterisk", algorithm=MD5, uri="sip:00@trunk1.netfuse.org", nonce="077112db", response="eca2f2ea3ae18207917559d1dd114c18"
Call-ID: e595ea6544af755b@10.10.1.4
CSeq: 57405 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) ---
Ignoring this INVITE request
trunk1*CLI>
<--- SIP read from UDP://87.81.167.157:5062 --->
ACK sip:00@trunk1.netfuse.org SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5062;branch=z9hG4bK94fe5bba64e1d271;rport
From: "Leo Brown" <sip:leo_kitchen@trunk1.netfuse.org>;tag=b3fc348eb47a6ce3
To: <sip:0@trunk1.netfuse.org>;tag=as60dfe957
Contact: <sip:leo_kitchen@87.81.167.157:5062;transport=udp>
Supported: path
Authorization: Digest username="leo_kitchen", realm="asterisk", algorithm=MD5, uri="sip:00@trunk1.netfuse.org", nonce="077112db", response="eca2f2ea3ae18207917559d1dd114c18"
Call-ID: e595ea6544af755b@10.10.1.4
CSeq: 57405 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) ---
Comments:By: Leo Brown (netfuse) 2009-06-25 07:19:18

Note from the two boxes, the key lines are:

1. Good
- INVITE sip:00@trunk1.netfuse.org SIP/2.0
- Using INVITE request as basis request - 8c0a8e3ebbf4d665@10.10.1.4

2. Bad
- INVITE sip:00@trunk1.netfuse.org SIP/2.0
- Ignoring this INVITE request

With exactly the same packets going across.
By: Leo Brown (netfuse) 2009-08-12 11:22:05

Am happy to test further as required.
By: Mark Michelson (mmichelson) 2009-08-20 13:04:45

So here's the deal for the "fast" case. The first transaction is one where an INVITE is sent to Asterisk, Asterisk requests authentication, and the phone ACKs.

Then, the phone sends an INVITE to '0' with CSeq 57405 and includes authentication information. Asterisk then sends back a 484. Now, here's where things get weird. The phone SHOULD be sending an ACK now, and then following that up with a new INVITE to '00' with CSeq 57406. Instead, no ACK is sent and the phone sends an INVITE to '00' with CSeq 57405. Because the CSeq is the same as the previous one, Asterisk thinks that this is a retransmission of the previous INVITE to '0'.

This is clearly a problem with the phone.
By: Leo Brown (netfuse) 2009-08-20 13:14:40

OK Give me a week for this, i am going to go back to Grandstream to see what they make of it.
By: Leif Madsen (lmadsen) 2009-09-17 15:07:06

Closed due to lack of feedback from the reporter. I'm not convinced this is exactly an issue with Asterisk though.