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:[email protected]>;tag=23bfef509d1f572f
To: <sip:[email protected]>;tag=as0c4f99e6
Call-ID: [email protected]
CSeq: 3180 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.215.146.162:5060;branch=z9hG4bKa6f748e3affb009b
From: "TEST" <sip:[email protected]>;tag=23bfef509d1f572f
To: <sip:[email protected]>;tag=as0c4f99e6
Contact: <sip:[email protected]:5060;transport=udp>
Supported: path
Proxy-Authorization: Digest username="4062", realm="asterisk", algorithm=MD5,
uri="sip:[email protected]", nonce="05e84442",
response="1f2b9e65c103a8b3b6973b77add91926"
Call-ID: [email protected]
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 [...@macro-dial:1] Macro("SIP/4062-08549df8", "hangupcall") in
new stack
-- Executing [...@macro-hangupcall:1] ResetCDR("SIP/4062-08549df8", "w") in
new stack
-- Executing [...@macro-hangupcall:2] NoCDR("SIP/4062-08549df8", "") in new
stack
-- Executing [...@macro-hangupcall:3] GotoIf("SIP/4062-08549df8",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [...@macro-hangupcall:6] GotoIf("SIP/4062-08549df8",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [...@macro-hangupcall:9] GotoIf("SIP/4062-08549df8",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [...@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:[email protected]>;tag=5a504797c7294815 To:
<sip:[email protected]>;tag=as6e9b4ae1 Call-ID:
[email protected] CSeq: 3180 INVITE User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported:
replaces Contact: <sip:[email protected]> 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:[email protected] SIP/2.0 Via: SIP/2.0/UDP
10.215.146.162:5060;branch=z9hG4bK772f38f747f7b80c From: "TEST"
<sip:[email protected]>;tag=5a504797c7294815 To:
<sip:[email protected]>;tag=as6e9b4ae1 Contact:
<sip:[email protected]:5060;transport=udp> Supported: path
Proxy-Authorization: Digest username="4062", realm="asterisk", algorithm=MD5,
uri="sip:[email protected]", nonce="613a5f53",
response="9914ddff3a8c46a8442841c426b98e98" Call-ID:
[email protected] 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:[email protected] SIP/2.0 Via: SIP/2.0/UDP
10.215.146.162:5060;branch=z9hG4bKb97ceeeceb3ee3dd From: "TEST"
<sip:[email protected]>;tag=5a504797c7294815 To:
<sip:[email protected]>;tag=as6e9b4ae1 Supported: path Proxy-Authorization:
Digest username="4062", realm="asterisk", algorithm=MD5,
uri="sip:[email protected]", nonce="613a5f53",
response="5ccaa2818cfcf70ce3a5c951caaec8ca" Call-ID:
[email protected] 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:[email protected]>;tag=5a504797c7294815 To:
<sip:[email protected]>;tag=as6e9b4ae1 Call-ID:
[email protected] 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
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