[Asterisk-Users] Re: Continuing Budgetone woes: asterisk was the culprit!!

[Brian Capouch]

Brian Capouch wrote:
> I have spent the morning on this project, still without success.
> 

When I saw the mail on the list tonight from "lists at uc9.net" it finally 
dawned on me to try a CVS-revert and see what happens.

It turns out that solved the problem--I can't say when exactly the bug 
showed up, but I did fresh CVS builds a couple of times in the past 
couple of days and they *have* the bug.

I'm currently running CVS-09/21/03-14:44:56 and it works just fine.

I'm not smart enough to spot the bug, nor to even elucidate it beyond 
the bounds of "it breaks Grandstream phones."  So I'll let someone else 
tell me whether/how to do a bug report.

I do know that in the future I will sure be quicker to suspect asterisk 
in this sort of an unexplained phenom; I lost two days on this, and 
nearly wore out the flash memory on my phone changing its configs and 
resetting it :-)

SIP trace of attempted call below in case it helps anyone.

Thx.

B.
> 
> ------------------------------------------------------------------------
> 
> Sip read: 
> INVITE sip:8 at 192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.21
> From: "BC IP Phone" <sip:btel at 192.168.1.10>;tag=c3cedeba-47c2-6790-8eb4-5b15010f6079
> To: <sip:8 at 192.168.1.10>
> Contact: <sip:btel at 192.168.1.21>
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c at 192.168.1.21
> CSeq: 39684 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: 257
> 
> v=0
> o=btel 0 0 IN IP4 192.168.1.21
> s=-
> c=IN IP4 192.168.1.21
> 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 192.168.1.21 : 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 - 269/0, 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 192.168.1.21
> From: "BC IP Phone" <sip:btel at 192.168.1.10>;tag=c3cedeba-47c2-6790-8eb4-5b15010f6079
> To: <sip:8 at 192.168.1.10>;tag=as5a4f2d09
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c at 192.168.1.21
> CSeq: 39684 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: 
> Proxy-Authenticate: Digest realm="asterisk", nonce="1ff7b5c9"
> Content-Length: 0
> 
> 
>  to 192.168.1.21:5060
> Sip read: 
> ACK sip:8 at 192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.21
> From: "BC IP Phone" <sip:btel at 192.168.1.10>;tag=c3cedeba-47c2-6790-8eb4-5b15010f6079
> To: <sip:8 at 192.168.1.10>;tag=as5a4f2d09
> Contact: <sip:btel at 192.168.1.21>
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c at 192.168.1.21
> CSeq: 39684 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:8 at 192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.21
> From: "BC IP Phone" <sip:btel at 192.168.1.10>;tag=6790947e-5b15-47c2-6079-8eb4e8bb010f
> To: <sip:8 at 192.168.1.10>
> Contact: <sip:btel at 192.168.1.21>
> Proxy-Authorization: DIGEST username="btel", realm="asterisk", algorithm=MD5, uri="sip:8 at 192.168.1.10", nonce="1ff7b5c9", response="c29fe4eab2affa88d79c91555a824c93"
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c at 192.168.1.21
> CSeq: 39685 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: 257
> 
> v=0
> o=btel 0 0 IN IP4 192.168.1.21
> s=-
> c=IN IP4 192.168.1.21
> 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 192.168.1.21 : 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 - 269/0, combined - 12
> Non-codec capabilities: us - 1, them - 0, combined - 0
> Looking for 8 in home
> list_route: hop: <sip:btel at 192.168.1.21>
> Transmitting (no NAT):
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.1.21
> From: "BC IP Phone" <sip:btel at 192.168.1.10>;tag=6790947e-5b15-47c2-6079-8eb4e8bb010f
> To: <sip:8 at 192.168.1.10>;tag=as2a2e797c
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c at 192.168.1.21
> CSeq: 39685 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:8 at 192.168.1.10>
> Content-Length: 0
> 
> 
>  to 192.168.1.21:5060
>     -- Executing VoiceMailMain2("SIP/btel-6234", "") in new stack
> We're at 192.168.1.10 port 6078
> 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 192.168.1.21
> From: "BC IP Phone" <sip:btel at 192.168.1.10>;tag=6790947e-5b15-47c2-6079-8eb4e8bb010f
> To: <sip:8 at 192.168.1.10>;tag=as2a2e797c
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c at 192.168.1.21
> CSeq: 39685 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:8 at 192.168.1.10>
> Content-Type: application/sdp
> Content-Length: 179
> 
> v=0
> o=root 13159 13159 IN IP4 192.168.1.10
> s=session
> c=IN IP4 192.168.1.10
> t=0 0
> m=audio 6078 RTP/AVP 3 0 8
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> 
>  to 192.168.1.21:5060
>     -- Playing 'vm-login'
> Retransmitting #1 (no NAT):
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.21
> From: "BC IP Phone" <sip:btel at 192.168.1.10>;tag=6790947e-5b15-47c2-6079-8eb4e8bb010f
> To: <sip:8 at 192.168.1.10>;tag=as2a2e797c
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c at 192.168.1.21
> CSeq: 39685 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:8 at 192.168.1.10>
> Content-Type: application/sdp
> Content-Length: 179
> 
> v=0
> o=root 13159 13159 IN IP4 192.168.1.10
> s=session
> c=IN IP4 192.168.1.10
> t=0 0
> m=audio 6078 RTP/AVP 3 0 8
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> 
>  to 192.168.1.21:5060
>