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<DIV><FONT face=Arial size=2>I just downloaded the newest version from CVS(<A
href="mailto:Tuesday@~7pm">Tuesday@~7pm</A>) 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.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2> -- Executing
Playback("SIP/budgtone-7ee9", "carried-away-by-monkeys") in new
stack<BR> -- Playing 'carried-away-by-monkeys' (language
'en')<BR> -- Executing Playback("SIP/budgtone-7ee9",
"lots-o-monkeys") in new stack<BR> -- Playing 'lots-o-monkeys'
(language 'en')<BR>WARNING[40966]: File chan_sip.c, Line 456 (retrans_pkt):
Maximum retries exceeded on call <A
href="mailto:d21f4608-1b1f-0a52-b657-2d9ca6239169@192.168.1.223">d21f4608-1b1f-0a52-b657-2d9ca6239169@192.168.1.223</A>
for seqno 1735 (Response)</FONT></DIV>
<DIV> </DIV><FONT face=Arial size=2>
<DIV><BR>With sip debug</DIV>
<DIV> </DIV>
<DIV>Sip read: <BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
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 </DIV>
<DIV> </DIV>
<DIV>12 headers, 13 lines</DIV>
<DIV> </DIV>
<DIV>Using latest request as basis request</DIV>
<DIV> </DIV>
<DIV>Sending to 192.168.1.223 : 5060 (non-NAT)</DIV>
<DIV> </DIV>
<DIV>Found audio format UNKN</DIV>
<DIV> </DIV>
<DIV>Found audio format ALAW</DIV>
<DIV> </DIV>
<DIV>Found audio format ULAW</DIV>
<DIV> </DIV>
<DIV>Found audio format UNKN</DIV>
<DIV> </DIV>
<DIV>Found audio format GSM</DIV>
<DIV> </DIV>
<DIV>Found audio format UNKN</DIV>
<DIV> </DIV>
<DIV>Found description format PCMU</DIV>
<DIV> </DIV>
<DIV>Found description format PCMA</DIV>
<DIV> </DIV>
<DIV>Found description format G723</DIV>
<DIV> </DIV>
<DIV>Found description format G729</DIV>
<DIV> </DIV>
<DIV>Found description format G726-32</DIV>
<DIV> </DIV>
<DIV>Found description format G728</DIV>
<DIV> </DIV>
<DIV>Capabilities: us - 524302, them - 285/0, combined - 12</DIV>
<DIV> </DIV>
<DIV>Non-codec capabilities: us - 1, them - 0, combined - 0</DIV>
<DIV> </DIV>
<DIV>Reliably Transmitting (no NAT):<BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
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 <BR> to 192.168.1.223:5060</DIV>
<DIV> </DIV>
<DIV>Sip read: <BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
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 </DIV>
<DIV> </DIV>
<DIV>11 headers, 0 lines</DIV>
<DIV> </DIV>
<DIV>Sip read: <BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
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 </DIV>
<DIV> </DIV>
<DIV>13 headers, 13 lines</DIV>
<DIV> </DIV>
<DIV>Using latest request as basis request</DIV>
<DIV> </DIV>
<DIV>Sending to 192.168.1.223 : 5060 (non-NAT)</DIV>
<DIV> </DIV>
<DIV>Found audio format UNKN</DIV>
<DIV> </DIV>
<DIV>Found audio format ALAW</DIV>
<DIV> </DIV>
<DIV>Found audio format ULAW</DIV>
<DIV> </DIV>
<DIV>Found audio format UNKN</DIV>
<DIV> </DIV>
<DIV>Found audio format GSM</DIV>
<DIV> </DIV>
<DIV>Found audio format UNKN</DIV>
<DIV> </DIV>
<DIV>Found description format PCMU</DIV>
<DIV> </DIV>
<DIV>Found description format PCMA</DIV>
<DIV> </DIV>
<DIV>Found description format G723</DIV>
<DIV> </DIV>
<DIV>Found description format G729</DIV>
<DIV> </DIV>
<DIV>Found description format G726-32</DIV>
<DIV> </DIV>
<DIV>Found description format G728</DIV>
<DIV> </DIV>
<DIV>Capabilities: us - 524302, them - 285/0, combined - 12</DIV>
<DIV> </DIV>
<DIV>Non-codec capabilities: us - 1, them - 0, combined - 0</DIV>
<DIV> </DIV>
<DIV>Looking for 9998 in default</DIV>
<DIV> </DIV>
<DIV>list_route: hop: <sip:budgtone@192.168.1.223></DIV>
<DIV> </DIV>
<DIV>Transmitting (no NAT):<BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER Contact: <sip:9998@192.168.1.2> Content-Length: 0
<BR> to 192.168.1.223:5060</DIV>
<DIV> </DIV>
<DIV> -- Executing Playback("SIP/budgtone-66e9",
"carried-away-by-monkeys") in new stack</DIV>
<DIV> </DIV>
<DIV>We're at 192.168.1.2 port 15592</DIV>
<DIV> </DIV>
<DIV>Answering with capability 2</DIV>
<DIV> </DIV>
<DIV>Answering with capability 4</DIV>
<DIV> </DIV>
<DIV>Answering with capability 8</DIV>
<DIV> </DIV>
<DIV>Reliably Transmitting (no NAT):<BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
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 <BR> to 192.168.1.223:5060</DIV>
<DIV> </DIV>
<DIV> -- Playing 'carried-away-by-monkeys' (language
'en')</DIV>
<DIV> </DIV>
<DIV>Retransmitting #1 (no NAT):<BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
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 <BR> to 192.168.1.223:5060</DIV>
<DIV> </DIV>
<DIV>Retransmitting #2 (no NAT):<BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
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 <BR> to 192.168.1.223:5060</DIV>
<DIV> </DIV>
<DIV> -- Executing Playback("SIP/budgtone-66e9",
"lots-o-monkeys") in new stack</DIV>
<DIV> </DIV>
<DIV> -- Playing 'lots-o-monkeys' (language 'en')</DIV>
<DIV> </DIV>
<DIV> -- Registered 'blah' (AUTHENTICATED) at
192.168.1.214:5036</DIV>
<DIV> </DIV>
<DIV>Retransmitting #3 (no NAT):<BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
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 <BR> to 192.168.1.223:5060</DIV>
<DIV> </DIV>
<DIV>Retransmitting #4 (no NAT):<BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
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 <BR> to 192.168.1.223:5060</DIV>
<DIV> </DIV>
<DIV>Retransmitting #5 (no NAT):<BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
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 <BR> to
192.168.1.223:5060<BR>WARNING[40966]: File chan_sip.c, Line 456 (retrans_pkt):
Maximum retries exceeded on call <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
for seqno 62160 (Response)</DIV>
<DIV> </DIV>
<DIV> == Spawn extension (default, 9998, 2) exited non-zero on
'SIP/budgtone-66e9'</DIV>
<DIV> </DIV>
<DIV>set_destination: Parsing <sip:budgtone@192.168.1.223> for
address/port to send to</DIV>
<DIV> </DIV>
<DIV>set_destination: set destination to 192.168.1.223, port 5060</DIV>
<DIV> </DIV>
<DIV>Reliably Transmitting:<BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to
192.168.1.223:5060</DIV>
<DIV> </DIV>
<DIV>Sip read: <BR>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: <A
href="mailto:fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223">fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223</A>
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 </DIV>
<DIV> </DIV>
<DIV>10 headers, 0 lines</DIV>
<DIV> </DIV>
<DIV>Message is BYE</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV><BR>Thanks for all your help<BR>
Will</FONT></DIV></BODY></HTML>