Grandstream GXP2000 and Incomplete() app

system · September 3, 2012, 10:15pm

Hi,

I’m facing an issue with a Grandstream GXP2000 configured for “early dial” and the Incomplete() application.

Firmware for the GXP2000 is 1.2.5.3, Asterisk is 10.7.0.

When using “early dial” with a “normal” dialplan with an ambiguous extension beginning, everything works as expected. For example, I have several 3 digit extensions starting with 9. So dialing 9 results in:

<--- SIP read from UDP:192.168.1.103:5060 --->
INVITE sip:9@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKed6049390c29cb0c;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f33ea13e26607b4f
To: <sip:9@asterisk.XXX:50060;user=phone>
Contact: <sip:110@192.168.1.103:5060;transport=udp;user=phone>
Supported: replaces, timer, path
P-Early-Media: Supported
Call-ID: d9c3457153b08389@192.168.1.103
CSeq: 62512 INVITE
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 405

v=0
o=110 8000 8000 IN IP4 192.168.1.103
s=SIP Call
c=IN IP4 192.168.1.103
t=0 0
m=audio 5054 RTP/AVP 8 0 4 18 2 97 9 3 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
--- (14 headers 19 lines) ---
Sending to 192.168.1.103:5060 (no NAT)
Using INVITE request as basis request - d9c3457153b08389@192.168.1.103
Found peer '110' for '110' from 192.168.1.103:5060

<--- Reliably Transmitting (no NAT) to 192.168.1.103:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKed6049390c29cb0c;received=192.168.1.103;rport=5060
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f33ea13e26607b4f
To: <sip:9@asterisk.XXX:50060;user=phone>;tag=as1d14521c
Call-ID: d9c3457153b08389@192.168.1.103
CSeq: 62512 INVITE
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="15aadfc1"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'd9c3457153b08389@192.168.1.103' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.103:5060 --->
ACK sip:9@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKed6049390c29cb0c;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f33ea13e26607b4f
To: <sip:9@asterisk.XXX:50060;user=phone>;tag=as1d14521c
Contact: <sip:110@192.168.1.103:5060;transport=udp;user=phone>
Supported: path
Call-ID: d9c3457153b08389@192.168.1.103
CSeq: 62512 ACK
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.103:5060 --->
INVITE sip:9@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKcd01334879d930c5;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f33ea13e26607b4f
To: <sip:9@asterisk.XXX:50060;user=phone>
Contact: <sip:110@192.168.1.103:5060;transport=udp;user=phone>
Supported: replaces, timer, path
P-Early-Media: Supported
Authorization: Digest username="110", realm="asterisk", algorithm=MD5, uri="sip:9@asterisk.XXX:50060;user=phone", nonce="15aadfc1", response="---"
Call-ID: d9c3457153b08389@192.168.1.103
CSeq: 62513 INVITE
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 405

v=0
o=110 8000 8001 IN IP4 192.168.1.103
s=SIP Call
c=IN IP4 192.168.1.103
t=0 0
m=audio 5054 RTP/AVP 8 0 4 18 2 97 9 3 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
--- (15 headers 19 lines) ---
Sending to 192.168.1.103:5060 (no NAT)
Using INVITE request as basis request - d9c3457153b08389@192.168.1.103
Found peer '110' for '110' from 192.168.1.103:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found audio description format G722 for ID 9
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.103:5054
Looking for 9 in client-sip-internal (domain asterisk.XXX)

<--- Reliably Transmitting (no NAT) to 192.168.1.103:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKcd01334879d930c5;received=192.168.1.103;rport=5060
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f33ea13e26607b4f
To: <sip:9@asterisk.XXX:50060;user=phone>;tag=as1d14521c
Call-ID: d9c3457153b08389@192.168.1.103
CSeq: 62513 INVITE
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'd9c3457153b08389@192.168.1.103' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.103:5060 --->
ACK sip:9@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKcd01334879d930c5;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f33ea13e26607b4f
To: <sip:9@asterisk.XXX:50060;user=phone>;tag=as1d14521c
Contact: <sip:110@192.168.1.103:5060;transport=udp;user=phone>
Supported: path
Authorization: Digest username="110", realm="asterisk", algorithm=MD5, uri="sip:9@asterisk.XXX:50060;user=phone", nonce="15aadfc1", response="---"
Call-ID: d9c3457153b08389@192.168.1.103
CSeq: 62513 ACK
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---

The sequence is INVITE -> 484 -> ACK

After the next digit, process restarts with new - now two digit - number and so on. All is well.

Now I want to go to more fancy things using the Incomplete() application. For testing purpose I defined an extension as follows:

exten => _6!,1,Incomplete(n)
 same =>      n,NoOp(What now?)

expecting I now can dial forever.

Unfortunately, the trace is now as follows:

<--- SIP read from UDP:192.168.1.103:5060 --->
INVITE sip:6@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK6a09bc3013026a02;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>
Contact: <sip:110@192.168.1.103:5060;transport=udp;user=phone>
Supported: replaces, timer, path
P-Early-Media: Supported
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19111 INVITE
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 405

v=0
o=110 8000 8000 IN IP4 192.168.1.103
s=SIP Call
c=IN IP4 192.168.1.103
t=0 0
m=audio 5086 RTP/AVP 8 0 4 18 2 97 9 3 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
--- (14 headers 19 lines) ---
Sending to 192.168.1.103:5060 (no NAT)
Using INVITE request as basis request - 49e9093cf03b5e76@192.168.1.103
Found peer '110' for '110' from 192.168.1.103:5060

<--- Reliably Transmitting (no NAT) to 192.168.1.103:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK6a09bc3013026a02;received=192.168.1.103;rport=5060
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>;tag=as17ff515d
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19111 INVITE
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1bad4b1d"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '49e9093cf03b5e76@192.168.1.103' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.103:5060 --->
ACK sip:6@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK6a09bc3013026a02;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>;tag=as17ff515d
Contact: <sip:110@192.168.1.103:5060;transport=udp;user=phone>
Supported: path
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19111 ACK
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.103:5060 --->
INVITE sip:6@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>
Contact: <sip:110@192.168.1.103:5060;transport=udp;user=phone>
Supported: replaces, timer, path
P-Early-Media: Supported
Authorization: Digest username="110", realm="asterisk", algorithm=MD5, uri="sip:6@asterisk.XXX:50060;user=phone", nonce="1bad4b1d", response="---"
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 INVITE
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 405

v=0
o=110 8000 8001 IN IP4 192.168.1.103
s=SIP Call
c=IN IP4 192.168.1.103
t=0 0
m=audio 5086 RTP/AVP 8 0 4 18 2 97 9 3 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
--- (15 headers 19 lines) ---
Sending to 192.168.1.103:5060 (no NAT)
Using INVITE request as basis request - 49e9093cf03b5e76@192.168.1.103
Found peer '110' for '110' from 192.168.1.103:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found audio description format G722 for ID 9
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.103:5086
Looking for 6 in client-sip-internal (domain asterisk.XXX)
list_route: hop: <sip:110@192.168.1.103:5060;transport=udp;user=phone>

<--- Transmitting (no NAT) to 192.168.1.103:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;received=192.168.1.103;rport=5060
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 INVITE
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6@192.168.1.105:50060>
Content-Length: 0


<------------>
    -- Executing [6@client-sip-internal:1] Incomplete("SIP/110-00000253", "n") in new stack

<--- Reliably Transmitting (no NAT) to 192.168.1.103:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;received=192.168.1.103;rport=5060
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>;tag=as0c1e6f7b
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 INVITE
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (client-sip-internal, 6, 1) exited INCOMPLETE on 'SIP/110-00000253'
    -- Sent into invalid extension '6' in context 'client-sip-internal' on SIP/110-00000253
    -- Executing [i@client-sip-internal:1] Playback("SIP/110-00000253", "invalid") in new stack
  == Spawn extension (client-sip-internal, i, 1) exited non-zero on 'SIP/110-00000253'
    -- Executing [h@client-sip-internal:1] Hangup("SIP/110-00000253", "") in new stack
  == Spawn extension (client-sip-internal, h, 1) exited non-zero on 'SIP/110-00000253'

<--- SIP read from UDP:192.168.1.103:5060 --->
ACK sip:6@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>;tag=as0c1e6f7b
Contact: <sip:110@192.168.1.103:5060;transport=udp;user=phone>
Supported: path
Authorization: Digest username="110", realm="asterisk", algorithm=MD5, uri="sip:6@asterisk.XXX:50060;user=phone", nonce="1bad4b1d", response="---"
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 ACK
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '49e9093cf03b5e76@192.168.1.103' Method: ACK

<--- SIP read from UDP:192.168.1.103:5060 --->
CANCEL sip:6@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>
Supported: path
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 CANCEL
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- Transmitting (no NAT) to 192.168.1.103:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;rport;received=192.168.1.103
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>;tag=as3c0bd612
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 CANCEL
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.1.103:5060 --->
CANCEL sip:6@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>
Supported: path
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 CANCEL
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- Transmitting (no NAT) to 192.168.1.103:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;rport;received=192.168.1.103
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>;tag=as4725558a
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 CANCEL
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.1.103:5060 --->
CANCEL sip:6@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>
Supported: path
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 CANCEL
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- Transmitting (no NAT) to 192.168.1.103:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;rport;received=192.168.1.103
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>;tag=as04a8768e
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 CANCEL
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.1.103:5060 --->
CANCEL sip:6@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>
Supported: path
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 CANCEL
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- Transmitting (no NAT) to 192.168.1.103:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;rport;received=192.168.1.103
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>;tag=as3c862a72
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 CANCEL
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.1.103:5060 --->
CANCEL sip:6@asterisk.XXX:50060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;rport
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>
Supported: path
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 CANCEL
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- Transmitting (no NAT) to 192.168.1.103:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKafbf02e760c0e6c0;rport;received=192.168.1.103
From: "Wohnung" <sip:110@asterisk.XXX:50060;user=phone>;tag=f6daf9eab6fd7950
To: <sip:6@asterisk.XXX:50060;user=phone>;tag=as72be4a5a
Call-ID: 49e9093cf03b5e76@192.168.1.103
CSeq: 19112 CANCEL
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

The sequence is now INVITE -> 100, 484 -> ACK.

After the 100 Trying, the Grandstream doesn’t accept any more digits if dialed. I hear DTMF, but no new INVITE is sent. When I finally give up and hang up, the Grandstream tries to cancel the call and Asterisk answers 481 saying there’s nothing to cancel.

Obviously, the only difference is the 100 Trying. So, MUST, SHOULD or MAY it be there, making it a fault of the Grandstream, or it MUST NOT be there, in which it would be a bug in Asterisk to open a ticket for. Or is it just me not being able to correctly configure Asterisk and/or interpreting the docs?

Any help would be highly appreciated.

Bye,
Christian

david55 · September 4, 2012, 10:16am

That’s a Grandstream problem. The 100 Trying is purely there to stop the device retransmitting INVITEs.

system · September 4, 2012, 1:48pm

Thank you for your assistance.

I now opened a ticket at Grandstream (though their ticket system seems to be usable with Internet Explorer only ) where I also refered to this thread.

For anyone who is interested in this topic I will post here the ticket outcome.

Bye,
Christian

system · September 19, 2012, 3:17pm

Update:

Grandstream could reproduce and confirm this issue. If I understood correctly, it’s now handed over to the development department and put in the queue for a next firmware update - if there should be any, since the GXP2000 reached “maintenance mode”.

I guess there isn’t a chance to suppress the 100 Trying in Asterisk, is there?

Bye,
Christian

david55 · September 19, 2012, 3:38pm

I don’t think there is any state associated with them, so you could remove them from the code.

This could result in redundant retransmissions of INVITEs, by the peers, and it could mean that the peer is unable to cancel a call setup (you can’t cancel until you have had some 1xx response (and you can’t cancel after a higher numbered one, as the call is either up, or gone).

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