I just downloaded the newest version from CVS(Tuesday@~7pm) and I am getting an
error whenever I call the asterisk box. I cannot here any audio on the budgtone.
This works fine with my pingtel phone and my sip 7960. Also if I call my Skinny
7960 it rings but I get that same error when I pick up. When the skinny phone
calls the Budgtone it works fine. I have 2 budgtone phones and it does this on
both of them. This worked fine before I installed the newest version of
asterisk.
    -- Executing Playback("SIP/budgtone-7ee9",
"carried-away-by-monkeys") in new stack
    -- Playing 'carried-away-by-monkeys' (language 'en')
    -- Executing Playback("SIP/budgtone-7ee9",
"lots-o-monkeys") in new stack
    -- Playing 'lots-o-monkeys' (language 'en')
WARNING[40966]: File chan_sip.c, Line 456 (retrans_pkt): Maximum retries
exceeded on call d21f4608-1b1f-0a52-b657-2d9ca6239169@192.168.1.223 for seqno
1735 (Response)
With sip debug
Sip read: 
INVITE sip:9998@192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.223 From:
"William Carlson"
<sip:budgtone@192.168.1.2>;tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To:
<sip:9998@192.168.1.2> Contact: <sip:budgtone@192.168.1.223>
Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62159 INVITE
User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow: INVITE, ACK,
CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type:
application/sdp Content-Length: 263  v=0 o=budgtone 0 0 IN IP4 192.168.1.223 s=-
c=IN IP4 192.168.1.223 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=ptime:20
a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18
G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000
12 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.223 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format ULAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 285/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.223 From:
"William Carlson"
<sip:budgtone@192.168.1.2>;tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To:
<sip:9998@192.168.1.2>;tag=as67b6f854 Call-ID:
fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62159 INVITE
User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:  Proxy-Authenticate: Digest realm="asterisk",
nonce="6c3e5732" Content-Length: 0
 to 192.168.1.223:5060
Sip read: 
ACK sip:9998@192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.223 From:
"William Carlson"
<sip:budgtone@192.168.1.2>;tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To:
<sip:9998@192.168.1.2>;tag=as67b6f854 Contact:
<sip:budgtone@192.168.1.223> Call-ID:
fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62159 ACK User-Agent:
Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE,
NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Length: 0
11 headers, 0 lines
Sip read: 
INVITE sip:9998@192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.223 From:
"William Carlson"
<sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To:
<sip:9998@192.168.1.2> Contact: <sip:budgtone@192.168.1.223>
Proxy-Authorization: DIGEST username="budgtone",
realm="asterisk", algorithm=MD5, uri="sip:9998@192.168.1.2",
nonce="6c3e5732",
response="4e90c985822b15d83f297e8c4fe80372" Call-ID:
fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE
User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow: INVITE, ACK,
CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type:
application/sdp Content-Length: 263  v=0 o=budgtone 0 0 IN IP4 192.168.1.223 s=-
c=IN IP4 192.168.1.223 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=ptime:20
a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18
G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000
13 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.223 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format ULAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 285/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for 9998 in default
list_route: hop: <sip:budgtone@192.168.1.223>
Transmitting (no NAT):
SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.223 From: "William
Carlson"
<sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To:
<sip:9998@192.168.1.2>;tag=as5481a27e Call-ID:
fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE
User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9998@192.168.1.2> Content-Length: 0
 to 192.168.1.223:5060
    -- Executing Playback("SIP/budgtone-66e9",
"carried-away-by-monkeys") in new stack
We're at 192.168.1.2 port 15592
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 192.168.1.223 From: "William Carlson"
<sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To:
<sip:9998@192.168.1.2>;tag=as5481a27e Call-ID:
fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE
User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9998@192.168.1.2> Content-Type: application/sdp
Content-Length: 176  v=0 o=root 7654 7654 IN IP4 192.168.1.2 s=session c=IN IP4
192.168.1.2 t=0 0 m=audio 15592 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0
PCMU/8000 a=rtpmap:8 PCMA/8000
 to 192.168.1.223:5060
    -- Playing 'carried-away-by-monkeys' (language 'en')
Retransmitting #1 (no NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson"
<sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To:
<sip:9998@192.168.1.2>;tag=as5481a27e Call-ID:
fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE
User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9998@192.168.1.2> Content-Type: application/sdp
Content-Length: 176  v=0 o=root 7654 7654 IN IP4 192.168.1.2 s=session c=IN IP4
192.168.1.2 t=0 0 m=audio 15592 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0
PCMU/8000 a=rtpmap:8 PCMA/8000
 to 192.168.1.223:5060
Retransmitting #2 (no NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson"
<sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To:
<sip:9998@192.168.1.2>;tag=as5481a27e Call-ID:
fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE
User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9998@192.168.1.2> Content-Type: application/sdp
Content-Length: 176  v=0 o=root 7654 7654 IN IP4 192.168.1.2 s=session c=IN IP4
192.168.1.2 t=0 0 m=audio 15592 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0
PCMU/8000 a=rtpmap:8 PCMA/8000
 to 192.168.1.223:5060
    -- Executing Playback("SIP/budgtone-66e9",
"lots-o-monkeys") in new stack
    -- Playing 'lots-o-monkeys' (language 'en')
    -- Registered 'blah' (AUTHENTICATED) at 192.168.1.214:5036
Retransmitting #3 (no NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson"
<sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To:
<sip:9998@192.168.1.2>;tag=as5481a27e Call-ID:
fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE
User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9998@192.168.1.2> Content-Type: application/sdp
Content-Length: 176  v=0 o=root 7654 7654 IN IP4 192.168.1.2 s=session c=IN IP4
192.168.1.2 t=0 0 m=audio 15592 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0
PCMU/8000 a=rtpmap:8 PCMA/8000
 to 192.168.1.223:5060
Retransmitting #4 (no NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson"
<sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To:
<sip:9998@192.168.1.2>;tag=as5481a27e Call-ID:
fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE
User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9998@192.168.1.2> Content-Type: application/sdp
Content-Length: 176  v=0 o=root 7654 7654 IN IP4 192.168.1.2 s=session c=IN IP4
192.168.1.2 t=0 0 m=audio 15592 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0
PCMU/8000 a=rtpmap:8 PCMA/8000
 to 192.168.1.223:5060
Retransmitting #5 (no NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson"
<sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To:
<sip:9998@192.168.1.2>;tag=as5481a27e Call-ID:
fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE
User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9998@192.168.1.2> Content-Type: application/sdp
Content-Length: 176  v=0 o=root 7654 7654 IN IP4 192.168.1.2 s=session c=IN IP4
192.168.1.2 t=0 0 m=audio 15592 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0
PCMU/8000 a=rtpmap:8 PCMA/8000
 to 192.168.1.223:5060
WARNING[40966]: File chan_sip.c, Line 456 (retrans_pkt): Maximum retries
exceeded on call fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 for seqno
62160 (Response)
  == Spawn extension (default, 9998, 2) exited non-zero on
'SIP/budgtone-66e9'
set_destination: Parsing <sip:budgtone@192.168.1.223> for address/port to
send to
set_destination: set destination to 192.168.1.223, port 5060
Reliably Transmitting:
BYE sip:budgtone@192.168.1.223 SIP/2.0 Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK4062184f From:
<sip:9998@192.168.1.2>;tag=as5481a27e To: "William Carlson"
<sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093
Contact: <sip:9998@192.168.1.2> Call-ID:
fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 102 BYE User-Agent:
Asterisk PBX Content-Length: 0   (no NAT) to 192.168.1.223:5060
Sip read: 
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4062184f From:
<sip:9998@192.168.1.2>;tag=as5481a27e To: "William Carlson"
<sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093
Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 102 BYE
User-Agent: Grandstream SIP UA 1.0.3.81 Contact:
<sip:budgtone@192.168.1.223> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY,
REFER, OPTIONS, INFO, SUBSCRIBE Content-Length: 0
10 headers, 0 lines
Message is BYE
Thanks for all your help
   Will
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20031105/55cdc2bd/attachment.htm