[Asterisk-Users] More Sip/Grandstream issues

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Sat, 27 Sep 2003 20:16:26 -0000

I just checkout the cvs code for asterisk......

when I use my grandstream phone (that worked on the old code that was
about 2 months old) I do not hear anything at all...

I get this error:
Sep 27 23:20:27 WARNING[1142127920]: File chan_sip.c, Line 444
(retrans_pkt): Maximum retries exceeded on call
0765c89e-9d67-3c0a-b9b9-2e7f3cd1d9ef@192.168.50.248 for seqno 58430
(Response)


here is my sip debug:

    -- Executing VoiceMailMain2("SIP/mlh-b787", "") in new
stack
We're at 192.168.50.1 port 27838
Answering with capability 2
Answering with capability 4
Answering with capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.248
From: "Michael Hess"
<sip:mlh@192.168.50.1>;tag=f3e33d4d-5431-b2d1-443e-0183d2cac6c7
To: <sip:8@192.168.50.1>;tag=as568b15d0
Call-ID: ccf9d2ca-982b-523b-89bb-10ce270b5847@192.168.50.248
CSeq: 53592 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8@192.168.50.1>
Content-Type: application/sdp
Content-Length: 178

v=0
o=root 1434 1434 IN IP4 192.168.50.1
s=session
c=IN IP4 192.168.50.1
t=0 0
m=audio 27838 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 192.168.50.248:5060
    -- Playing 'vm-login'
Retransmitting #1 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.248
From: "Michael Hess"
<sip:mlh@192.168.50.1>;tag=f3e33d4d-5431-b2d1-443e-0183d2cac6c7
To: <sip:8@192.168.50.1>;tag=as568b15d0
Call-ID: ccf9d2ca-982b-523b-89bb-10ce270b5847@192.168.50.248
CSeq: 53592 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8@192.168.50.1>
Content-Type: application/sdp
Content-Length: 178

v=0
o=root 1434 1434 IN IP4 192.168.50.1
s=session
c=IN IP4 192.168.50.1
t=0 0
m=audio 27838 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
 ACKA
 to 192.168.50.248:5060
Retransmitting #2 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.248
From: "Michael Hess"
<sip:mlh@192.168.50.1>;tag=f3e33d4d-5431-b2d1-443e-0183d2cac6c7
To: <sip:8@192.168.50.1>;tag=as568b15d0
Call-ID: ccf9d2ca-982b-523b-89bb-10ce270b5847@192.168.50.248
CSeq: 53592 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8@192.168.50.1>
Content-Type: application/sdp
Content-Length: 178

v=0
o=root 1434 1434 IN IP4 192.168.50.1
s=session
c=IN IP4 192.168.50.1
t=0 0
m=audio 27838 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
 ACKA
 to 192.168.50.248:5060
Retransmitting #3 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.248
From: "Michael Hess"
<sip:mlh@192.168.50.1>;tag=f3e33d4d-5431-b2d1-443e-0183d2cac6c7
To: <sip:8@192.168.50.1>;tag=as568b15d0
Call-ID: ccf9d2ca-982b-523b-89bb-10ce270b5847@192.168.50.248
CSeq: 53592 INVITE
User-Agent: Asterisk PBX