Non-codec capabilities (dtmf): us - 0x1(telephone-event),

2 msg	lightweight prepaid app using Dial andextention...
20 msg	QOS for outgoing SIP calls
3 msg	extenspy and chanspy
3 msg	chan_zap error 1.4.19 tone duration
17 msg	Drag and Drop transfer application
2 msg	Problem with hints (1.4.19)
12 msg	Chanspy on Asterisk 1.4.19
4 msg	Using Chanspy
5 msg	PSTN to SIP
2 msg	asterisk trunk
8 msg	Best Click-to-call client
1 msg	Callerid Error
4 msg	Busy (congestion) signal and cell phones
11 msg	DUNDi and SIP

Non-codec capabilities (dtmf): us - 0x1(telepho...
\ broadband Voice (16 Apr 2008)
. \ Steve Totaro (16 Apr 2008)

1 msg	Version FIOS MWI Detection - asterisk-1.6-beta7
3 msg	Hangup conundrum with RxFAX
10 msg	Two annoying bugs of asterisk ( sip in use anda...
5 msg	Simple queue announcements
3 msg	Problem with B410P

Subject:	Non-codec capabilities (dtmf): us - 0x1(telephone-event),
Group:	Asterisk-users
From:	broadband Voice
Date:	16 Apr 2008

We have two servers but looks like G729 issues. Works fine on the old server
and not sure if it is T1 related. See SIP Debug. Any experiences to share.
Thanks

---
Newark1*CLI>
<--- SIP read from 194.xx.Xx.Xx:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 76.xx.xx.xx:5060;branch=xxxxK784d2637;rport
From: "Cell Phone DC" <sip:202xxxxxxx>;tag=as04819ca3
To: <sip:xx>;tag=xx
Contact: sip:251xxxxxxxx:5060
Call-ID: xxx
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACKBYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 198

v=0
o=xxxxxx 12xxxxx 12xxxx IN IP4 62.xx.xx.xx
s=SIP Call
c=IN IP4 62.xx.xx.xxx
t=0 0
m=audio 8786 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20

<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 62.xx.xx.xx:8786
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 62.xx.xx.xx:8786
-- SIP/Voicetrading-08e1ce18 is making progress passing it to Zap/5-1

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