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Brian Capouch

2003-Sep-27 13:16 UTC

[Asterisk-Users] Continuing Budgetone woes

I have spent the morning on this project, still without success.

Summary: Yesterday I inadvertently unplugged my Grandstream phone.  I 
might add I did a rebuild of my s/w from CVS at the same time.  Since 
then, the Budgetone seems to talk SIP just fine, but the RTP being sent 
to it by asterisk "doesn't make any sound."

It was suggested I do a factory reset of the phone, which I did.

I have traced until I'm blue in the face--the SIP part looks normal and 
the asterisk server sends its audio stream just like it's supposed to, 
but I can't hear a thing.  Calls to NuFone and other providers proceed 
normally until the start of the RTP stream, then stop cold.

I am suspecting codec compatibility, but ain't smart enough to figure 
out from the attached trace whether my hunch might be correct.

Could someone take a quick gander at the RTP negotiation in this trace 
and see if there is a smoking gun--I dialed "8" on the Budgetone 
(192.168.1.21) to get voicemail on the asterisk box (192.168.1.10).

Thanks.

B.
-------------- next part --------------
Sip read: 
INVITE sip:8@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.21
From: "BC IP Phone"
<sip:btel@192.168.1.10>;tag=c3cedeba-47c2-6790-8eb4-5b15010f6079
To: <sip:8@192.168.1.10>
Contact: <sip:btel@192.168.1.21>
Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@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@192.168.1.10>;tag=c3cedeba-47c2-6790-8eb4-5b15010f6079
To: <sip:8@192.168.1.10>;tag=as5a4f2d09
Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@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@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.21
From: "BC IP Phone"
<sip:btel@192.168.1.10>;tag=c3cedeba-47c2-6790-8eb4-5b15010f6079
To: <sip:8@192.168.1.10>;tag=as5a4f2d09
Contact: <sip:btel@192.168.1.21>
Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@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@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.21
From: "BC IP Phone"
<sip:btel@192.168.1.10>;tag=6790947e-5b15-47c2-6079-8eb4e8bb010f
To: <sip:8@192.168.1.10>
Contact: <sip:btel@192.168.1.21>
Proxy-Authorization: DIGEST username="btel",
realm="asterisk", algorithm=MD5, uri="sip:8@192.168.1.10",
nonce="1ff7b5c9",
response="c29fe4eab2affa88d79c91555a824c93"
Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@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@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@192.168.1.10>;tag=6790947e-5b15-47c2-6079-8eb4e8bb010f
To: <sip:8@192.168.1.10>;tag=as2a2e797c
Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@192.168.1.21
CSeq: 39685 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8@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@192.168.1.10>;tag=6790947e-5b15-47c2-6079-8eb4e8bb010f
To: <sip:8@192.168.1.10>;tag=as2a2e797c
Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@192.168.1.21
CSeq: 39685 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8@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@192.168.1.10>;tag=6790947e-5b15-47c2-6079-8eb4e8bb010f
To: <sip:8@192.168.1.10>;tag=as2a2e797c
Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@192.168.1.21
CSeq: 39685 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8@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

Brian Capouch

2003-Sep-27 22:24 UTC

head link

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

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@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@192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.21
> From: "BC IP Phone"
<sip:btel@192.168.1.10>;tag=c3cedeba-47c2-6790-8eb4-5b15010f6079
> To: <sip:8@192.168.1.10>
> Contact: <sip:btel@192.168.1.21>
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@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@192.168.1.10>;tag=c3cedeba-47c2-6790-8eb4-5b15010f6079
> To: <sip:8@192.168.1.10>;tag=as5a4f2d09
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@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@192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.21
> From: "BC IP Phone"
<sip:btel@192.168.1.10>;tag=c3cedeba-47c2-6790-8eb4-5b15010f6079
> To: <sip:8@192.168.1.10>;tag=as5a4f2d09
> Contact: <sip:btel@192.168.1.21>
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@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@192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.21
> From: "BC IP Phone"
<sip:btel@192.168.1.10>;tag=6790947e-5b15-47c2-6079-8eb4e8bb010f
> To: <sip:8@192.168.1.10>
> Contact: <sip:btel@192.168.1.21>
> Proxy-Authorization: DIGEST username="btel",
realm="asterisk", algorithm=MD5, uri="sip:8@192.168.1.10",
nonce="1ff7b5c9",
response="c29fe4eab2affa88d79c91555a824c93"
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@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@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@192.168.1.10>;tag=6790947e-5b15-47c2-6079-8eb4e8bb010f
> To: <sip:8@192.168.1.10>;tag=as2a2e797c
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@192.168.1.21
> CSeq: 39685 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:8@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@192.168.1.10>;tag=6790947e-5b15-47c2-6079-8eb4e8bb010f
> To: <sip:8@192.168.1.10>;tag=as2a2e797c
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@192.168.1.21
> CSeq: 39685 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:8@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@192.168.1.10>;tag=6790947e-5b15-47c2-6079-8eb4e8bb010f
> To: <sip:8@192.168.1.10>;tag=as2a2e797c
> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@192.168.1.21
> CSeq: 39685 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:8@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
>

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