asterisk users - Apr 2008 - Non-codec capabilities (dtmf): us - 0x1 (telephone-event),

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 at
76.xx.xx.xx>;tag=as04819ca3
To: <sip:xx>;tag=xx
Contact: sip:251xxxxxxxx at 194.xx.xx.XX:5060
Call-ID: xxx at 76.x.x.x
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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On Wed, Apr 16, 2008 at 9:10 AM, broadband Voice
<broadbandvoice at gmail.com> wrote:
> 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 at
76.xx.xx.xx>;tag=as04819ca3
> To: <sip:xx>;tag=xx
> Contact: sip:251xxxxxxxx at 194.xx.xx.XX:5060
>  Call-ID: xxx at 76.x.x.x
> 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
Looks to be OK to me but you have negotiated Ulaw not G729.

Thanks,
Steve Totaro