Dear Sir,
I'm sending them Session Progress as you can see in the attached log
fle...Please let me know if they ahve any reason to not sending DTMF to me
Regards
On Fri, Oct 3, 2008 at 6:54 PM, Alex Balashov <[EMAIL PROTECTED]>wrote:
> 200 OK is a SIP response indicating the successful establishment of an
> INVITE transaction.
>
> I can think of no reason why you would not be sending a 200 OK to your
> provider unless you are failing to Answer() the call in your dial plan
> and are instead sending them early media (183 Session in Progress).
>
> A packet capture would be most helpful.
>
> michel freiha wrote:
>
> > Dear All,
> >
> > I have a DTMF problem with VOxBone, the company that provide us the DID
> > numbers...Sometimes they sent us DTMF packets and sometimes not...
> > VoxBone said asterisk is not sending back OK message to their Gateway
> > that's why they are not sending us the DTMF packets...How to force
> > Asterisk server to reply back by sending OK message?
> >
> > Regards
> >
> >
> >
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
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> >
> > asterisk-users mailing list
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>
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
<--- SIP read from 81.201.82.39:5060 --->
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
From: "anonymous" ;tag=70665
To:
Via: SIP/2.0/UDP
81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7
Max-Forwards: 69
Content-Type: application/sdp
Contact:
User-Agent: Vox Callcontrol
Content-Length: 311
v=0
o=root 16790 16790 IN IP4 81.201.82.23
s=session
c=IN IP4 81.201.82.23
t=0 0
m=audio 11564 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<->
--- (11 headers 15 lines) ---
Sending to 81.201.82.39 : 5060 (no NAT)
Using INVITE request as basis request - [EMAIL PROTECTED]
Found peer 'sip_proxy1'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 81.201.82.23:11564
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/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 81.201.82.23:11564
Looking for 155469877445 in stations (domain Asterisk_IP)
list_route: hop:
localhost*CLI>
<--- Transmitting (no NAT) to 81.201.82.39:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7;received=81.201.82.39
From: "anonymous" ;tag=70665
To:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
<
<--- Transmitting (no NAT) to 81.201.82.39:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7;received=81.201.82.39
From: "anonymous" ;tag=70665
To: ;tag=as78e4c405
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 637 637 IN IP4 Asterisk_IP
s=session
c=IN IP4 Asterisk_IP
t=0 0
m=audio 17750 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
localhost*CLI>
<--- SIP read from 81.201.82.39:5060 --->
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
From: "anonymous" ;tag=70665
To:
Via: SIP/2.0/UDP
81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7
Max-Forwards: 69
User-Agent: Vox Callcontrol
Content-Length: 0
<->
--- (9 headers 0 lines) ---
Sending to 81.201.82.39 : 5060 (no
