[Asterisk-Users] DTMF broken

2004-05-09 Thread Mark Elkins 
Some CVS upgrade in the last day or two has broken the recognition of
DTMF eg in Voicemail. I'm running the latest CVS as of now. I'm getting
the error...

*CLI -- Executing VoiceMailMain(SIP/phone1-e0dd, ) in new stack
-- Playing 'vm-login' (language 'en')
**Here I push a button**
May  9 18:26:18 WARNING[98311]: chan_sip.c:5027 receive_info: Unable to
retrieve DTMF signal from INFO message from
[EMAIL PROTECTED]

By re-installing an older (cvs checkout -r v1-0_stable asterisk) version
- everything works fine again... thats with NO config changes at all..
Has someone removed some support for the transporting of DTMF (eg,
info?) - I am using... dtmfmode=info in sip.conf with BudgeTone-100's

(sent with absolutely no signatures or attachments)


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Description: This is a digitally signed message part

Re: [Asterisk-Users] DTMF broken

2004-05-09 Thread brian k. west 
What firmware you have on that BT101?  And yes gnupg or what ever you use to
sign your message did produce the attachemnt on this last one too.

bkw

- Original Message - 
From: Mark Elkins [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, May 09, 2004 12:23 PM
Subject: [Asterisk-Users] DTMF broken



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Re: [Asterisk-Users] DTMF broken

2004-05-09 Thread Olle E. Johansson 
Mark,
Could you please add a SIP debug message with the SIP INFO?
/O
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Re: [Asterisk-Users] DTMF broken

2004-05-09 Thread Mark Elkins 
On Sun, 2004-05-09 at 20:48, Olle E. Johansson wrote:
 Mark,
 Could you please add a SIP debug message with the SIP INFO?

I've done a debug with a working asterisk (V1.0) and the non-working
asterisk. The trace is attached.  :-)(debug - ascii text)

When you say SIP INFO - what else are you asking for???
If its one of the 'sip show' commands - which one, and at what instance
of time?

-- 
  .  . ___. .__  Posix Systems - Sth Africa
 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

This is a debug trace of Asterisk v1-0_stable - I'm dialing '310' which in 
extensions.conf looks like..
; 310 = Access Voicemail - with full prompting
exten = 310,1,VoicemailMain()

I'm hanging up after 'dialing' 203
... the 'bad' one follows after

*CLI sip debug
SIP Debugging Enabled
*CLI 

Sip read: 
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK5550a5fc35b24bcb
From: Phone One sip:[EMAIL PROTECTED];user=phone;tag=63f98f4e24e20f2f
To: sip:[EMAIL PROTECTED];user=phone
Contact: sip:[EMAIL PROTECTED];user=phone
Call-ID: [EMAIL PROTECTED]
CSeq: 2408 INVITE
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 345

v=0
o=phone1 8000 8000 IN IP4 160.124.48.121
s=SIP Call
c=IN IP4 160.124.48.121
t=0 0
m=audio 5004 RTP/AVP 98 0 8 18 9 4 2 15
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=ptime:40

12 headers, 16 lines
Using latest request as basis request
Sending to 160.124.48.121 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found audio format ULAW
Found audio format GSM
Found audio format UNKN
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G722
Found description format G723
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 1309/0, 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 160.124.48.121;branch=z9hG4bK5550a5fc35b24bcb
From: Phone One sip:[EMAIL PROTECTED];user=phone;tag=63f98f4e24e20f2f
To: sip:[EMAIL PROTECTED];user=phone;tag=as3564c06e
Call-ID: [EMAIL PROTECTED]
CSeq: 2408 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Proxy-Authenticate: Digest realm=asterisk, nonce=6d4d7372
Content-Length: 0


 to 160.124.48.121:5060


Sip read: 
ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK5550a5fc35b24bcb
From: Phone One sip:[EMAIL PROTECTED];user=phone;tag=63f98f4e24e20f2f
To: sip:[EMAIL PROTECTED];user=phone;tag=as3564c06e
Contact: sip:[EMAIL PROTECTED];user=phone
Call-ID: [EMAIL PROTECTED]
CSeq: 2408 ACK
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


11 headers, 0 lines


Sip read: 
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKf13dbe7ea5fc60a6
From: Phone One sip:[EMAIL PROTECTED];user=phone;tag=63f98f4e24e20f2f
To: sip:[EMAIL PROTECTED];user=phone
Contact: sip:[EMAIL PROTECTED];user=phone
Proxy-Authorization: DIGEST username=phone1, realm=asterisk, algorithm=MD5, 
uri=sip:[EMAIL PROTECTED]
a;user=phone, nonce=6d4d7372, response=0142fb85eda2d7497992a0149d78e828
Call-ID: [EMAIL PROTECTED]
CSeq: 2409 INVITE
User-Agent: Grandstream BT100 1.0.4.63
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 345

v=0
o=phone1 8000 8000 IN IP4 160.124.48.121
s=SIP Call
c=IN IP4 160.124.48.121
t=0 0
m=audio 5004 RTP/AVP 98 0 8 18 9 4 2 15
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=ptime:40

13 headers, 16 lines
Using latest request as basis request
Sending to 160.124.48.121 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found audio format ULAW
Found audio format GSM
Found audio format UNKN
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G722
Found description format G723
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 1309/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0

Re: [Asterisk-Users] DTMF broken

2004-05-09 Thread Mark Elkins 
On Sun, 2004-05-09 at 21:39, brian k. west wrote:
 What firmware you have on that BT101?  And yes gnupg or what ever you use to
 sign your message did produce the attachemnt on this last one too.

OK the gnuPG is off.. :-(

Product Model:BT100
Software Version: Program--1.0.4.63 Bootloader--1.0.0.16 HTML--1.0.0.30
VOC--1.0.0.5

-- 
  .  . ___. .__  Posix Systems - Sth Africa
 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

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