[Asterisk-Users] SIP broken for budgtone.

William Carlson wcarlson at w0ss.com
Wed Nov 5 04:12:05 MST 2003 

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 at 192.168.1.223 for seqno 1735 (Response)


With sip debug

Sip read: 
INVITE sip:9998 at 192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone at 192.168.1.2>;tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To: <sip:9998 at 192.168.1.2> Contact: <sip:budgtone at 192.168.1.223> Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 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 at 192.168.1.2>;tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To: <sip:9998 at 192.168.1.2>;tag=as67b6f854 Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 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 at 192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone at 192.168.1.2>;tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To: <sip:9998 at 192.168.1.2>;tag=as67b6f854 Contact: <sip:budgtone at 192.168.1.223> Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 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 at 192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone at 192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998 at 192.168.1.2> Contact: <sip:budgtone at 192.168.1.223> Proxy-Authorization: DIGEST username="budgtone", realm="asterisk", algorithm=MD5, uri="sip:9998 at 192.168.1.2", nonce="6c3e5732", response="4e90c985822b15d83f297e8c4fe80372" Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 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 at 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 at 192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998 at 192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998 at 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 at 192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998 at 192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998 at 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 at 192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998 at 192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998 at 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 at 192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998 at 192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998 at 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 at 192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998 at 192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998 at 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 at 192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998 at 192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998 at 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 at 192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998 at 192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998 at 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 at 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 at 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 at 192.168.1.223 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4062184f From: <sip:9998 at 192.168.1.2>;tag=as5481a27e To: "William Carlson" <sip:budgtone at 192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 Contact: <sip:9998 at 192.168.1.2> Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 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 at 192.168.1.2>;tag=as5481a27e To: "William Carlson" <sip:budgtone at 192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52 at 192.168.1.223 CSeq: 102 BYE User-Agent: Grandstream SIP UA 1.0.3.81 Contact: <sip:budgtone at 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
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