Hi,
I have several GXP2000 phones which used to work fine with Asterisk 1.2.
However, after upgrading to Asterisk 1.4.21.2, whenever I initiate a call from a
GXP2000, it gets dropped after 20 seconds exactly.
I have "early dial" enabled on the GXP2000 and
"pedantic=yes" on the server. If I disable "early dial", all
works well ("early dial" or "overlap dial" is used when the
server supports "484 address incomplete" replies).
Can someone please let me know if it's an Asterisk or a Grandstream bug?
Basically, I think that my problem is that I'm getting a "481 Call
leg/transaction does not exist".
A sip debug ip <GXP2000 IP> yields the following (GXP2000 extension 4062
at 10.215.146.162 calls softphone extension 4053 at 10.215.144.48 via Asterisk
1.4 at 10.215.147.112):
Retransmitting #6 (NAT) to 10.215.146.162:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.215.146.162:5060;branch=z9hG4bK70d7f269b5551fce;received=10.215.146.162
From: "TEST" <sip:4062 at pbx.voip.local>;tag=23bfef509d1f572f
To: <sip:4053 at pbx.voip.local>;tag=as0c4f99e6
Call-ID: 3b1f444665f731f7 at 10.215.146.162
CSeq: 3180 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:4053 at 10.215.147.112>
Content-Type: application/sdp
Content-Length: 243
v=0
o=root 12813 12813 IN IP4 10.215.147.112
s=session
c=IN IP4 10.215.147.112
t=0 0
m=audio 13290 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
inf-voip2*CLI>
<--- SIP read from 10.215.146.162:5060 --->
ACK sip:4053 at 10.215.147.112 SIP/2.0
Via: SIP/2.0/UDP 10.215.146.162:5060;branch=z9hG4bKa6f748e3affb009b
From: "TEST" <sip:4062 at pbx.voip.local>;tag=23bfef509d1f572f
To: <sip:4053 at pbx.voip.local>;tag=as0c4f99e6
Contact: <sip:4062 at 10.215.146.162:5060;transport=udp>
Supported: path
Proxy-Authorization: Digest username="4062",
realm="asterisk", algorithm=MD5, uri="sip:4053 at
pbx.voip.local", nonce="05e84442",
response="1f2b9e65c103a8b3b6973b77add91926"
Call-ID: 3b1f444665f731f7 at 10.215.146.162
CSeq: 3180 ACK
User-Agent: Grandstream GXP2000 1.1.6.44
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
== Spawn extension (macro-dial, s, 7) exited non-zero on
'SIP/4062-08549df8' in macro 'dial'
== Spawn extension (macro-dial, s, 7) exited non-zero on
'SIP/4062-08549df8' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 7) exited non-zero on
'SIP/4062-08549df8'
-- Executing [h at macro-dial:1] Macro("SIP/4062-08549df8",
"hangupcall") in new stack
-- Executing [s at macro-hangupcall:1]
ResetCDR("SIP/4062-08549df8", "w") in new stack
-- Executing [s at macro-hangupcall:2] NoCDR("SIP/4062-08549df8",
"") in new stack
-- Executing [s at macro-hangupcall:3] GotoIf("SIP/4062-08549df8",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s at macro-hangupcall:6] GotoIf("SIP/4062-08549df8",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s at macro-hangupcall:9] GotoIf("SIP/4062-08549df8",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s at macro-hangupcall:11]
Hangup("SIP/4062-08549df8", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/4062-08549df8' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/4062-08549df8'
A syslog snippet of the GXP2000 is as follows:
Mar 9 10:02:30 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] SIPReceive(750, Account1):
SIP/2.0 200 OK Via: SIP/2.0/UDP
10.215.146.162:5060;branch=z9hG4bK8a4f132fa37f09b7;received=10.215.146.162
From: "TEST" <sip:4062 at pbx.voip.local>;tag=5a504797c7294815
To: <sip:4053 at pbx.voip.local>;tag=as6e9b4ae1 Call-ID: 881609a50827e9d7
at 10.215.146.162 CSeq: 3180 INVITE User-Agent: Asterisk PBX Allow: INVITE,
ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces
Contact: <sip:4053 at 10.215.147.112> Content-Type: application/sdp
Content-Length: 243 v=0 o=root 12813 12813 IN IP4 10.215.147.112 s=session
c=IN IP4 10.215.147.112 t=0 0 m=audio 16296 RTP/AVP 3 101 a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -
a=ptime:20 a=sendrecv
Mar 9 10:02:30 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] Received SIP message: 200
Mar 9 10:02:30 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] SIP dialog matched to channel 0
Mar 9 10:02:30 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] Session Info: Payload-Type=3,
Frames/Packet=1, DTMF=101
Mar 9 10:02:30 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] RTP session starts. Channel: 0
Local RTP port: 5032 Remote RTP endpoint: 10.215.147.112:16296
Mar 9 10:02:30 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] Send SIP message: ACK To
10.215.147.112:5060, sip_handle: 0x0052F0AA
Mar 9 10:02:30 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] sip_len: 717, sip_handle:
0x0052F0AA, ACK sip:4053 at 10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP
10.215.146.162:5060;branch=z9hG4bK772f38f747f7b80c From: "TEST"
<sip:4062 at pbx.voip.local>;tag=5a504797c7294815 To: <sip:4053 at
pbx.voip.local>;tag=as6e9b4ae1 Contact: <sip:4062 at
10.215.146.162:5060;transport=udp> Supported: path Proxy-Authorization:
Digest username="4062", realm="asterisk", algorithm=MD5,
uri="sip:4053 at pbx.voip.local", nonce="613a5f53",
response="9914ddff3a8c46a8442841c426b98e98" Call-ID: 881609a50827e9d7
at 10.215.146.162 CSeq: 3180 ACK User-Agent: Grandstream GXP2000 1.1.6.44
Max-Forwards: 70 Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
Mar 9 10:02:37 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] 2009-03-09 10:02:48 [Status]-ON
HOOK
Mar 9 10:02:37 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] Send SIP message: BYE To
10.215.147.112:5060, sip_handle: 0x0052F0AA
Mar 9 10:02:37 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] sip_len: 653, sip_handle:
0x0052F0AA, BYE sip:4053 at 10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP
10.215.146.162:5060;branch=z9hG4bKb97ceeeceb3ee3dd From: "TEST"
<sip:4062 at pbx.voip.local>;tag=5a504797c7294815 To: <sip:4053 at
pbx.voip.local>;tag=as6e9b4ae1 Supported: path Proxy-Authorization: Digest
username="4062", realm="asterisk", algorithm=MD5,
uri="sip:4053 at 10.215.147.112", nonce="613a5f53",
response="5ccaa2818cfcf70ce3a5c951caaec8ca" Call-ID: 881609a50827e9d7
at 10.215.146.162 CSeq: 3181 BYE User-Agent: Grandstream GXP2000 1.1.6.44
Max-Forwards: 70 Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
Mar 9 10:02:37 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] 2009-03-09 10:02:48 Tone stop
(0)
Mar 9 10:02:37 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] 2009-03-09 10:02:48 LCD
Callmode: CALLMODE_NULL
Mar 9 10:02:37 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] 2009-03-09 10:02:48 Voc mode
(0): CALLMODE_NULL
Mar 9 10:02:37 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] 2009-03-09 10:02:48 Aud path
(0): AUD_PATH_NULL
Mar 9 10:02:37 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] SIPReceive(468, Account1):
SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP
10.215.146.162:5060;branch=z9hG4bKb97ceeeceb3ee3dd;received=10.215.146.162
From: "TEST" <sip:4062 at pbx.voip.local>;tag=5a504797c7294815
To: <sip:4053 at pbx.voip.local>;tag=as6e9b4ae1 Call-ID: 881609a50827e9d7
at 10.215.146.162 CSeq: 3181 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces
Content-Length: 0
Mar 9 10:02:37 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] Received SIP message: 481
Mar 9 10:02:37 10.215.146.162 GS_LOG:
[00:0B:82:19:AE:0B][000][9620000512B][0101062C] SIP dialog matched to channel 0
