Rajkumar S
2006-Oct-05 11:58 UTC
[asterisk-users] No voice for when using Playback and background
Hi, I am using 1.2.12.1 (actually was using 1.2.11, and upgraded) it's connected to a Cisco ATA 188. The phones connected to ATA can register to * and two phones connected to ATA can call each other. I can hear Music On Hold, when called using the following fragment exten => 6000,1,Answer exten => 6000,2,MusicOnHold() But the Playback and Background does not work, ie I cannot hear any thing. exten => 200,1,Playback(tt-allbusy) exten => 200,n,Playback(moo2) The sip.conf fragment for ATA Phone is [100] type=friend username=100 secret=password canreinvite=no host=dynamic dtmfmode=rfc2833 context = sip nat=1 Actually this was working couple of days back, the last modification done was to install zaptel and libpri. I have looked far and wide in google,but nothing came up. Any help to fix this will be much appreciated. raj
Mojo with Horan & Company, LLC
2006-Oct-05 14:46 UTC
[asterisk-users] No voice for when using Playback and background
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) Moj
Rajkumar S
2006-Oct-05 22:16 UTC
Fwd: [asterisk-users] No voice for when using Playback and background
On 10/5/06, Mojo with Horan & Company, LLC <mojo@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@192.168.9.224;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.9.230:5060 From: <sip:100@192.168.9.224;user=phone>;tag=3810654101 To: <sip:200@192.168.9.224;user=phone> Call-ID: 2005103074@192.168.9.230 CSeq: 1 INVITE Contact: <sip:100@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@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@192.168.9.224;user=phone>;tag=3810654101 To: <sip:200@192.168.9.224;user=phone>;tag=as43f3d7b7 Call-ID: 2005103074@192.168.9.230 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:200@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@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@192.168.9.224;user=phone>;tag=3810654101 To: <sip:200@192.168.9.224;user=phone>;tag=as43f3d7b7 Call-ID: 2005103074@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@192.168.9.224;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.9.230:5060 From: <sip:100@192.168.9.224;user=phone>;tag=3810654101 To: <sip:200@192.168.9.224;user=phone> Call-ID: 2005103074@192.168.9.230 CSeq: 2 INVITE Contact: <sip:100@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@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@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@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@192.168.9.224;user=phone>;tag=3810654101 To: <sip:200@192.168.9.224;user=phone> Call-ID: 2005103074@192.168.9.230 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:200@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@192.168.9.224;user=phone>;tag=3810654101 To: <sip:200@192.168.9.224;user=phone>;tag=as35a40f82 Call-ID: 2005103074@192.168.9.230 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:200@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@192.168.9.224 SIP/2.0 Via: SIP/2.0/UDP 192.168.9.230:5060 From: <sip:100@192.168.9.224;user=phone>;tag=3810654101 To: <sip:200@192.168.9.224;user=phone>;tag=as35a40f82 Call-ID: 2005103074@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@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@192.168.29.30 CSeq: 1718 OPTIONS From: <sip:1001@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@192.168.9.224>;tag=755247969947 To: <sip:192.168.9.224>;tag=as26f85f68 Call-ID: 43877BD0-5AC1-4404-9703-10AA66FF280F@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@192.168.29.30' The call is alive at this point. raj