Fwd: [asterisk-users] No voice for when using Playback and background

[Rajkumar S]

On 10/5/06, Mojo with Horan & Company, LLC <mojo at horanappraisals.com> wrote:
> See if adding an answer line helps:
>
> Rajkumar S wrote:
> > exten => 200,1,Playback(tt-allbusy)
> > exten => 200,n,Playback(moo2)
>
> change to:
>
> exten => 200,1,Answer
> exten => 200,n,Playback(tt-allbusy)
> exten => 200,n,Playback(moo2)

Nope, Infact I had tried this before posting to the list. The full sip debug is:

<-- SIP read from 192.168.9.230:5060:
INVITE sip:200 at 192.168.9.224;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.9.230:5060
From: <sip:100 at 192.168.9.224;user=phone>;tag=3810654101
To: <sip:200 at 192.168.9.224;user=phone>
Call-ID: 2005103074 at 192.168.9.230
CSeq: 1 INVITE
Contact: <sip:100 at 192.168.9.230:5060;user=phone;transport=udp>
User-Agent: Cisco ATA 188  v2.16.1 ata18x (030709a)
Expires: 300
Content-Length: 246
Content-Type: application/sdp

v=0
o=100 8904 8904 IN IP4 192.168.9.230
s=ATA186 Call
c=IN IP4 192.168.9.230
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (11 headers 11 lines)---
Using INVITE request as basis request - 2005103074 at 192.168.9.230
Sending to 192.168.9.230 : 5060 (non-NAT)
Reliably Transmitting (NAT) to 192.168.9.230:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.9.230:5060;received=192.168.9.230
From: <sip:100 at 192.168.9.224;user=phone>;tag=3810654101
To: <sip:200 at 192.168.9.224;user=phone>;tag=as43f3d7b7
Call-ID: 2005103074 at 192.168.9.230
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:200 at 192.168.9.224>
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="453e6aef"
Content-Length: 0


<-- SIP read from 192.168.9.230:5060:
ACK sip:200 at 192.168.9.224;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.9.230:5060;received=192.168.9.230
From: <sip:100 at 192.168.9.224;user=phone>;tag=3810654101
To: <sip:200 at 192.168.9.224;user=phone>;tag=as43f3d7b7
Call-ID: 2005103074 at 192.168.9.230
CSeq: 1 ACK
User-Agent: Cisco ATA 188  v2.16.1 ata18x (030709a)
Content-Length: 0


--- (8 headers 0 lines)---

<-- SIP read from 192.168.9.230:5060:
INVITE sip:200 at 192.168.9.224;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.9.230:5060
From: <sip:100 at 192.168.9.224;user=phone>;tag=3810654101
To: <sip:200 at 192.168.9.224;user=phone>
Call-ID: 2005103074 at 192.168.9.230
CSeq: 2 INVITE
Contact: <sip:100 at 192.168.9.230:5060;user=phone;transport=udp>
User-Agent: Cisco ATA 188  v2.16.1 ata18x (030709a)
Proxy-Authorization: Digest
username="100",realm="asterisk",nonce="453e6aef",uri="sip:200 at 192.168.9.224",response="4f0cfbdda408c879f8ac15bd27bcc02c"
Expires: 300
Content-Length: 246
Content-Type: application/sdp

v=0
o=100 8906 8906 IN IP4 192.168.9.230
s=ATA186 Call
c=IN IP4 192.168.9.230
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (12 headers 11 lines)---
Using INVITE request as basis request - 2005103074 at 192.168.9.230
Sending to 192.168.9.230 : 5060 (NAT)
Found user '100'
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.9.230:16384
Found description format PCMU
Found description format G723
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0xd
(g723|ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 200 in sip (domain 192.168.9.224;user=phone)
list_route: hop: <sip:100 at 192.168.9.230:5060;user=phone;transport=udp>
Transmitting (NAT) to 192.168.9.230:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.9.230:5060;received=192.168.9.230
From: <sip:100 at 192.168.9.224;user=phone>;tag=3810654101
To: <sip:200 at 192.168.9.224;user=phone>
Call-ID: 2005103074 at 192.168.9.230
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:200 at 192.168.9.224>
Content-Length: 0


---
    -- Executing Playback("SIP/100-081b28b8", "tt-allbusy") in new stack
We're at 192.168.9.224 port 14652
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.9.230:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.9.230:5060;received=192.168.9.230
From: <sip:100 at 192.168.9.224;user=phone>;tag=3810654101
To: <sip:200 at 192.168.9.224;user=phone>;tag=as35a40f82
Call-ID: 2005103074 at 192.168.9.230
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:200 at 192.168.9.224>
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 14808 14808 IN IP4 192.168.9.224
s=session
c=IN IP4 192.168.9.224
t=0 0
m=audio 14652 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Playing 'tt-allbusy' (language 'en')

<-- SIP read from 192.168.9.230:5060:
ACK sip:200 at 192.168.9.224 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.230:5060
From: <sip:100 at 192.168.9.224;user=phone>;tag=3810654101
To: <sip:200 at 192.168.9.224;user=phone>;tag=as35a40f82
Call-ID: 2005103074 at 192.168.9.230
CSeq: 2 ACK
User-Agent: Cisco ATA 188  v2.16.1 ata18x (030709a)
Proxy-Authorization: Digest
username="100",realm="asterisk",nonce="453e6aef",uri="sip:200 at 192.168.9.224",response="4f0cfbdda408c879f8ac15bd27bcc02c"
Content-Length: 0


--- (9 headers 0 lines)---

<-- SIP read from 192.168.29.30:5060:
OPTIONS sip:192.168.9.224 SIP/2.0
Via: SIP/2.0/UDP
192.168.29.30;rport;branch=z9hG4bKc0a81d1e000000104525e5140000769800001425
Content-Length: 0
Call-ID: 43877BD0-5AC1-4404-9703-10AA66FF280F at 192.168.29.30
CSeq: 1718 OPTIONS
From: <sip:1001 at 192.168.9.224>;tag=755247969947
Max-Forwards: 70
To: <sip:192.168.9.224>


--- (8 headers 0 lines)---
Looking for s in sip (domain 192.168.9.224)
Transmitting (no NAT) to 192.168.29.30:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.29.30;rport;branch=z9hG4bKc0a81d1e000000104525e5140000769800001425;received=192.168.29.30

From: <sip:1001 at 192.168.9.224>;tag=755247969947
To: <sip:192.168.9.224>;tag=as26f85f68
Call-ID: 43877BD0-5AC1-4404-9703-10AA66FF280F at 192.168.29.30
CSeq: 1718 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.9.224>
Accept: application/sdp
Content-Length: 0


---
Destroying call '43877BD0-5AC1-4404-9703-10AA66FF280F at 192.168.29.30'

The call is alive at this point.

raj