asterisk users - Feb 2009 - DTMF

IVR Number :17275691533

When I try it from xlite configuring my provider directly, it works
perfectly.

When I try to dial out from dialer , it doesnt work.

[sip8]
type=peer
username=user
fromuser=user
authuser=user
secret=password
host=8.14.146.111
nat=no
canreinvite=yes
insecure=very
disallow=all
allow=g729
allow=ulaw
context=default
dtmfmode=rfc2833


What cld be the reason ?
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Is your soft phone also using rfc2833 for DTMF mode? 

--Tim 

----- "David @ULC" wrote: 
> 
> IVR Number :17275691533 
> When I try it from xlite configuring my provider directly, it works
perfectly.
> 
> When I try to dial out from dialer , it doesnt work. 
> [sip8] 
> type=peer 
> username=user 
> fromuser=user 
> authuser=user 
> secret=password 
> host=8.14.146.111 
> nat=no 
> canreinvite=yes 
> insecure=very 
> disallow=all 
> allow=g729 
> allow=ulaw 
> context=default 
> dtmfmode=rfc2833 
> What cld be the reason ? 
> 
> 
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On 19 Feb 2009, at 20:48, David @ULC wrote:
> IVR Number :17275691533
>
> When I try it from xlite configuring my provider directly, it works  
> perfectly.
>
> When I try to dial out from dialer , it doesnt work.
>
> [sip8]
> type=peer
> username=user
> fromuser=user
> authuser=user
> secret=password
> host=8.14.146.111
> nat=no
> canreinvite=yes
> insecure=very
> disallow=all
> allow=g729
> allow=ulaw
> context=default
> dtmfmode=rfc2833
>
>
> What cld be the reason ?
Anything. Give us some debug output (sip debug would be perfect).

S

--- (12 headers 0 lines) ---
Sending to 192.168.0.50 : 12714 (NAT)
Transmitting (NAT) to 192.168.0.50:12714:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.50:12714
;branch=z9hG4bK-d87543-930550325154e53d-1--d87543-;received=192.168.0.50;rport=12714
From: "cc106"<sip:cc106 at 192.168.0.2 <sip%3Acc106 at
192.168.0.2>>;tag=7f1cff22
To: "817275691533"<sip:817275691533 at
192.168.0.2<sip%3A817275691533 at 192.168.0.2>
>;tag=as02559696
Call-ID: NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:817275691533 at 192.168.0.2 <sip%3A817275691533 at
192.168.0.2>>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing



---
Scheduling destruction of call
'617ad67d47db8e4a2155fcd51d1089ff at 59.xxx.xx.xx' in 32000 ms
set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for
address/port to send to
set_destination: set destination to 8.14.xxx.xxx, port 5060
Reliably Transmitting (no NAT) to 8.14.xxx.xxx:5060:
BYE sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK59c0212a;rport
From: "cc106" <sip:fiddialer at 59.xxx.xx.xx>;tag=as3f9466a7
To: <sip:17275691533 at 8.14.xxx.xxx>;tag=1902000923108720995156225
Call-ID: 617ad67d47db8e4a2155fcd51d1089ff at 59.xxx.xx.xx
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Spawn extension (default, 817275691533, 2) exited non-zero on
'SIP/cc106-b7a1a9d0'
    -- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi://
127.0.0.1:4577/call_log") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi://
127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12)")
in new stack
    -- AGI Script agi://
127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12)
completed, returning 0
Destroying call 'NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.'
vicidialnow*CLI>
<-- SIP read from 8.14.xxx.xxx:5060:
SIP/2.0 200 OK
CSeq: 102 INVITE
Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK3a111ef4;rport
From: "V0219160007000134649" <sip:fiddialer at
59.xxx.xx.xx>;tag=as79fae976
Call-ID: 1525f0ef1e787bed51ed1ef119adb1fa at 59.xxx.xx.xx
To: <sip:16785588539 at 8.14.xxx.xxx>;tag=1902000923098720982816221
Contact: <sip:8.14.xxx.xxx:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 225

v=0
o=VoipSwitch 7220 7220 IN IP4 8.14.xxx.xxx
s=VoipSIP
i=Audio Session
c=IN IP4 8.14.xxx.xxx
t=0 0
m=audio 6220 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (9 headers 11 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 8.14.xxx.xxx:6220
Found description format G729
Found description format telephone-event
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0
(nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:8.14.xxx.xxx:5060;transport=udp>
set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for
address/port to send to
set_destination: set destination to 8.14.xxx.xxx, port 5060
Transmitting (no NAT) to 8.14.xxx.xxx:5060:
ACK sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK6eef7893;rport
From: "V0219160007000134649" <sip:fiddialer at
59.xxx.xx.xx>;tag=as79fae976
To: <sip:16785588539 at 8.14.xxx.xxx>;tag=1902000923098720982816221
Contact: <sip:fiddialer at 59.xxx.xx.xx>
Call-ID: 1525f0ef1e787bed51ed1ef119adb1fa at 59.xxx.xx.xx
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
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