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Hold music not playing on MSS


dlynton
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After upgrading to the latest version of PBXNSIP, we are no longer getting hold music when we do a supervised transfer. As you can see in the call log below, pbxnsip is not attempting to play moh.wav, but it does say "call hold from trunk". We haven't changed anything in our application, just upgraded to the latest version of pbxnsip. Any ideas?

 

 

REGISTER sip:callcentric.com SIP/2.0

Via: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-2fbc56286f9c2e41e5b5a6024332fdf6;rport

From: "CallCentricSanta" <sip:17772557741@callcentric.com>;tag=41990

To: "CallCentricSanta" <sip:17772557741@callcentric.com>

Call-ID: 9enn0ykw@pbx

CSeq: 27448 REGISTER

Max-Forwards: 70

Contact: <sip:17772557741@192.168.54.103:5060;transport=udp;line=1679091c>;+sip.instance="<urn:uuid:63e9e564-771f-4789-9477-0fa6b0f68aa7>"

User-Agent: pbxnsip-PBX/2.1.6.2448

Expires: 3600

Content-Length: 0

 

 

[9] 2008/03/05 11:31:17: SIP Rx udp:204.11.192.23:5080:

SIP/2.0 200 Ok

v: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-2fbc56286f9c2e41e5b5a6024332fdf6;rport=5060

f: "CallCentricSanta" <sip:17772557741@callcentric.com>;tag=41990

t: "CallCentricSanta" <sip:17772557741@callcentric.com>

i: 9enn0ykw@pbx

CSeq: 27448 REGISTER

m: <sip:17772557741@192.168.54.103:5060;transport=udp;line=1679091c>;+sip.instance="<urn:uuid:63e9e564-771f-4789-9477-0fa6b0f68aa7>";expires=97

l: 0

 

 

[8] 2008/03/05 11:31:33: Found interface on 192.168.54.103 with netmask 255.255.255.0

[8] 2008/03/05 11:31:33: Found interface on 127.0.0.1 with netmask 255.0.0.0

[9] 2008/03/05 11:32:05: Resolve 163: url sip:callcentric.com

[9] 2008/03/05 11:32:05: Resolve 163: naptr callcentric.com

[9] 2008/03/05 11:32:05: Resolve 163: srv tls _sips._tcp.callcentric.com

[9] 2008/03/05 11:32:05: Resolve 163: srv tcp _sip._tcp.callcentric.com

[9] 2008/03/05 11:32:05: Resolve 163: srv udp _sip._udp.callcentric.com

[9] 2008/03/05 11:32:05: Resolve 163: a udp alpha1.callcentric.com 5080

[9] 2008/03/05 11:32:05: Resolve 163: udp 204.11.192.22 5080

[9] 2008/03/05 11:32:05: Resolve 163: a udp alpha2.callcentric.com 5080

[9] 2008/03/05 11:32:05: Resolve 163: udp 204.11.192.23 5080

[9] 2008/03/05 11:32:05: Resolve 163: a udp callcentric.com 5060

[9] 2008/03/05 11:32:05: Resolve 163: udp 204.11.192.23 5060

[8] 2008/03/05 11:32:05: Trunk 6 (callcentric) has outbound proxy udp:204.11.192.22:5080 udp:204.11.192.23:5060 udp:204.11.192.23:5080

[9] 2008/03/05 11:32:05: Resolve 164: udp 204.11.192.23 5080

[9] 2008/03/05 11:32:05: SIP Tx udp:204.11.192.23:5080:

REGISTER sip:callcentric.com SIP/2.0

Via: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-5c7846da9d261398d53f498c45f07a0e;rport

From: "CallCentricSanta" <sip:17772557741@callcentric.com>;tag=41990

To: "CallCentricSanta" <sip:17772557741@callcentric.com>

Call-ID: 9enn0ykw@pbx

CSeq: 27450 REGISTER

Max-Forwards: 70

Contact: <sip:17772557741@192.168.54.103:5060;transport=udp;line=1679091c>;+sip.instance="<urn:uuid:63e9e564-771f-4789-9477-0fa6b0f68aa7>"

User-Agent: pbxnsip-PBX/2.1.6.2448

Expires: 3600

Content-Length: 0

 

 

[9] 2008/03/05 11:32:05: SIP Rx udp:204.11.192.23:5080:

SIP/2.0 200 Ok

v: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-5c7846da9d261398d53f498c45f07a0e;rport=5060

f: "CallCentricSanta" <sip:17772557741@callcentric.com>;tag=41990

t: "CallCentricSanta" <sip:17772557741@callcentric.com>

i: 9enn0ykw@pbx

CSeq: 27450 REGISTER

m: <sip:17772557741@192.168.54.103:5060;transport=udp;line=1679091c>;+sip.instance="<urn:uuid:63e9e564-771f-4789-9477-0fa6b0f68aa7>";expires=98

l: 0

 

 

[9] 2008/03/05 11:32:12: SIP Rx udp:192.168.54.100:5060:

INVITE sip:7135599056@192.168.54.103 SIP/2.0

Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK708f672c;rport

From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224

To: <sip:7135599056@192.168.54.103>

Contact: <sip:2814610506@192.168.54.100>

Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 05 Mar 2008 17:49:53 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 265

 

v=0

o=root 5779 5779 IN IP4 192.168.54.100

s=session

c=IN IP4 192.168.54.100

t=0 0

m=audio 10858 RTP/AVP 3 0 8 101

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

 

[7] 2008/03/05 11:32:12: UDP: Opening socket on port 56516

[7] 2008/03/05 11:32:12: UDP: Opening socket on port 56517

[5] 2008/03/05 11:32:12: Identify trunk (IP address/port and domain match) 5

[9] 2008/03/05 11:32:12: Resolve 165: aaaa udp 192.168.54.100 5060

[9] 2008/03/05 11:32:12: Resolve 165: a udp 192.168.54.100 5060

[9] 2008/03/05 11:32:12: Resolve 165: udp 192.168.54.100 5060

[9] 2008/03/05 11:32:12: SIP Tx udp:192.168.54.100:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK708f672c;rport=5060

From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224

To: <sip:7135599056@192.168.54.103>;tag=00b1880f9c

Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100

CSeq: 102 INVITE

Content-Length: 0

 

 

[6] 2008/03/05 11:32:12: Sending RTP for 1a92846a480027bf329359bc5a096045@192.168.54.100#00b1880f9c to 192.168.54.100:10858

[5] 2008/03/05 11:32:12: Trunk asterisk outbound sends call to 7135599056

[7] 2008/03/05 11:32:12: Calling extension 7135599056

[6] 2008/03/05 11:32:12: Redirecting to external voicemail account 7135599056 destination sip:011997135599056@192.168.54.103

[9] 2008/03/05 11:32:12: Dialplan: Evaluating !^(7132234676)@.*!sip:\1@\r;user=phone!i against 011997135599056@192.168.54.103

[9] 2008/03/05 11:32:12: Dialplan: Evaluating !^01199([0-9]*)@.*!sip:\1@\r;user=phone!i against 011997135599056@192.168.54.103

[5] 2008/03/05 11:32:12: Dialplan default: Match 011997135599056@192.168.54.103 to <sip:7135599056@192.168.54.200;user=phone> on trunk MS Speech Server

[5] 2008/03/05 11:32:12: Using "DELESANDRI T " <sip:2814610506@192.168.54.103> as redirect from

[5] 2008/03/05 11:32:12: Charge user 7135599056 for redirecting calls

[8] 2008/03/05 11:32:12: Play audio_moh/noise.wav

[7] 2008/03/05 11:32:12: UDP: Opening socket on port 58934

[7] 2008/03/05 11:32:12: UDP: Opening socket on port 58935

[9] 2008/03/05 11:32:12: Resolve 166: url sip:192.168.54.200:5060;transport=tcp

[9] 2008/03/05 11:32:12: Resolve 166: a tcp 192.168.54.200 5060

[9] 2008/03/05 11:32:12: Resolve 166: tcp 192.168.54.200 5060

[9] 2008/03/05 11:32:12: SIP Tx tcp:192.168.54.200:5060:

INVITE sip:7135599056@192.168.54.200;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.54.103:2335;branch=z9hG4bK-1bdb15e9e72d3681cef849f25fc0b0c5;rport

From: "DELESANDRI T " <sip:2814610506@192.168.54.103>;tag=5140

To: <sip:7135599056@192.168.54.200;user=phone>

Call-ID: f89e1cc4@pbx

CSeq: 883 INVITE

Max-Forwards: 70

Contact: <sip:2814610506@192.168.54.103:2335;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2448

Diversion: <tel:7135599056>;reason=unavailable;screen=no;privacy=off

P-Preferred-Identity: <sip:7135599056@192.168.54.103>

Content-Type: application/sdp

Content-Length: 220

 

v=0

o=- 35388 35388 IN IP4 192.168.54.103

s=-

c=IN IP4 192.168.54.103

t=0 0

m=audio 58934 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[9] 2008/03/05 11:32:12: Resolve 167: aaaa udp 192.168.54.100 5060

[9] 2008/03/05 11:32:12: Resolve 167: a udp 192.168.54.100 5060

[9] 2008/03/05 11:32:12: Resolve 167: udp 192.168.54.100 5060

[9] 2008/03/05 11:32:12: SIP Tx udp:192.168.54.100:5060:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK708f672c;rport=5060

From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224

To: <sip:7135599056@192.168.54.103>;tag=00b1880f9c

Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100

CSeq: 102 INVITE

Contact: <sip:7135599056@192.168.54.103:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2448

Content-Type: application/sdp

Content-Length: 220

 

v=0

o=- 63166 63166 IN IP4 192.168.54.103

s=-

c=IN IP4 192.168.54.103

t=0 0

m=audio 56516 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:5060:

SIP/2.0 100 Trying

FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140

TO: <sip:7135599056@192.168.54.200;user=phone>

CSEQ: 883 INVITE

CALL-ID: f89e1cc4@pbx

VIA: SIP/2.0/TCP 192.168.54.103:2335;branch=z9hG4bK-1bdb15e9e72d3681cef849f25fc0b0c5;rport

CONTENT-LENGTH: 0

 

 

[9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:5060:

SIP/2.0 302 Moved Temporarily

FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140

TO: <sip:7135599056@192.168.54.200;user=phone>;tag=f4d3cd3c6c

CSEQ: 883 INVITE

CALL-ID: f89e1cc4@pbx

VIA: SIP/2.0/TCP 192.168.54.103:2335;branch=z9hG4bK-1bdb15e9e72d3681cef849f25fc0b0c5;rport

CONTACT: <sip:7135599056@192.168.54.200:2257;user=phone;transport=Tcp;maddr=192.168.54.200;x-mss-call-id=f89e1cc4%40pbx>

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 2008/03/05 11:32:12: Call f89e1cc4@pbx#5140: Clear last INVITE

[9] 2008/03/05 11:32:12: Resolve 168: url sip:7135599056@192.168.54.200;user=phone

[9] 2008/03/05 11:32:12: Resolve 168: udp 192.168.54.200 5060

[9] 2008/03/05 11:32:12: SIP Tx udp:192.168.54.200:5060:

ACK sip:7135599056@192.168.54.200;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-1bdb15e9e72d3681cef849f25fc0b0c5;rport

From: "DELESANDRI T " <sip:2814610506@192.168.54.103>;tag=5140

To: <sip:7135599056@192.168.54.200;user=phone>;tag=f4d3cd3c6c

Call-ID: f89e1cc4@pbx

CSeq: 883 ACK

Max-Forwards: 70

Contact: <sip:2814610506@192.168.54.103:5060;transport=udp>

P-Preferred-Identity: <sip:7135599056@192.168.54.103>

Content-Length: 0

 

 

[5] 2008/03/05 11:32:12: Redirecting call

[9] 2008/03/05 11:32:12: Resolve 169: aaaa tcp 192.168.54.200 2257

[9] 2008/03/05 11:32:12: Resolve 169: a tcp 192.168.54.200 2257

[9] 2008/03/05 11:32:12: Resolve 169: tcp 192.168.54.200 2257

[9] 2008/03/05 11:32:12: SIP Tx tcp:192.168.54.200:2257:

INVITE sip:7135599056@192.168.54.200:2257;user=phone;transport=Tcp;maddr=192.168.54.200;x-mss-call-id=f89e1cc4%40pbx SIP/2.0

Via: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-758cb24853e9a80d8f297be22b1b6852;rport

From: "DELESANDRI T " <sip:2814610506@192.168.54.103>;tag=5140

To: <sip:7135599056@192.168.54.200;user=phone>

Call-ID: f89e1cc4@pbx

CSeq: 884 INVITE

Max-Forwards: 70

Contact: <sip:2814610506@192.168.54.103:2336;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2448

Diversion: <tel:7135599056>;reason=unavailable;screen=no;privacy=off

P-Preferred-Identity: <sip:7135599056@192.168.54.103>

Content-Type: application/sdp

Content-Length: 220

 

v=0

o=- 35388 35388 IN IP4 192.168.54.103

s=-

c=IN IP4 192.168.54.103

t=0 0

m=audio 58934 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[8] 2008/03/05 11:32:12: UDP: recvfrom receives ICMP message

[5] 2008/03/05 11:32:12: Connection refused on udp:192.168.54.200:5060

[6] 2008/03/05 11:32:12: Could not determine destination address on 168

[9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:2257:

SIP/2.0 100 Trying

FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140

TO: <sip:7135599056@192.168.54.200;user=phone>

CSEQ: 884 INVITE

CALL-ID: f89e1cc4@pbx

VIA: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-758cb24853e9a80d8f297be22b1b6852;rport

CONTENT-LENGTH: 0

 

 

[9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:2257:

SIP/2.0 180 Ringing

FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140

TO: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf

CSEQ: 884 INVITE

CALL-ID: f89e1cc4@pbx

VIA: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-758cb24853e9a80d8f297be22b1b6852;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[8] 2008/03/05 11:32:12: Play audio_en/ringback.wav

[9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:2257:

SIP/2.0 200 OK

FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140

TO: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf

CSEQ: 884 INVITE

CALL-ID: f89e1cc4@pbx

VIA: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-758cb24853e9a80d8f297be22b1b6852;rport

CONTACT: <sip:vtspeech07.dc.voicetarget.com:2257;transport=Tcp;maddr=192.168.54.200>;automata

CONTENT-LENGTH: 198

CONTENT-TYPE: application/sdp

ALLOW: UPDATE

SERVER: RTCC/3.0.0.0

ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

 

v=0

o=- 0 0 IN IP4 192.168.54.200

s=Microsoft Speech Server session

c=IN IP4 192.168.54.200

t=0 0

m=audio 13440 RTP/AVP 0 8 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

[7] 2008/03/05 11:32:12: Call f89e1cc4@pbx#5140: Clear last INVITE

[7] 2008/03/05 11:32:12: Set packet length to 20

[6] 2008/03/05 11:32:12: Sending RTP for f89e1cc4@pbx#5140 to 192.168.54.200:13440

[9] 2008/03/05 11:32:12: Resolve 170: aaaa tcp 192.168.54.200 2257

[9] 2008/03/05 11:32:12: Resolve 170: a tcp 192.168.54.200 2257

[9] 2008/03/05 11:32:12: Resolve 170: tcp 192.168.54.200 2257

[9] 2008/03/05 11:32:12: SIP Tx tcp:192.168.54.200:2257:

ACK sip:vtspeech07.dc.voicetarget.com:2257;transport=Tcp;maddr=192.168.54.200 SIP/2.0

Via: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-a0f9d3e52482a7033ffd2563d08a6970;rport

From: "DELESANDRI T " <sip:2814610506@192.168.54.103>;tag=5140

To: <sip:7135599056@192.168.54.200;user=phone>;tag=73d97a1bbf

Call-ID: f89e1cc4@pbx

CSeq: 884 ACK

Max-Forwards: 70

Contact: <sip:2814610506@192.168.54.103:2336;transport=tcp>

P-Preferred-Identity: <sip:7135599056@192.168.54.103>

Content-Length: 0

 

 

[7] 2008/03/05 11:32:12: Determine pass-through mode after receiving response

[9] 2008/03/05 11:32:12: Resolve 171: aaaa udp 192.168.54.100 5060

[9] 2008/03/05 11:32:12: Resolve 171: a udp 192.168.54.100 5060

[9] 2008/03/05 11:32:12: Resolve 171: udp 192.168.54.100 5060

[9] 2008/03/05 11:32:12: SIP Tx udp:192.168.54.100:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK708f672c;rport=5060

From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224

To: <sip:7135599056@192.168.54.103>;tag=00b1880f9c

Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100

CSeq: 102 INVITE

Contact: <sip:7135599056@192.168.54.103:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2448

Content-Type: application/sdp

Content-Length: 220

 

v=0

o=- 63166 63166 IN IP4 192.168.54.103

s=-

c=IN IP4 192.168.54.103

t=0 0

m=audio 56516 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/03/05 11:32:12: f89e1cc4@pbx#5140: RTP pass-through mode

[7] 2008/03/05 11:32:12: 1a92846a480027bf329359bc5a096045@192.168.54.100#00b1880f9c: RTP pass-through mode

[9] 2008/03/05 11:32:12: SIP Rx udp:192.168.54.100:5060:

ACK sip:7135599056@192.168.54.103:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK5784477d;rport

From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224

To: <sip:7135599056@192.168.54.103>;tag=00b1880f9c

Contact: <sip:2814610506@192.168.54.100>

Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0

 

 

[9] 2008/03/05 11:32:39: SIP Rx tcp:192.168.54.200:2257:

INVITE sip:2814610506@192.168.54.103:2336;transport=tcp SIP/2.0

FROM: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf

TO: <sip:2814610506@192.168.54.103>;tag=5140

CSEQ: 1 INVITE

CALL-ID: f89e1cc4@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.54.200:2257;branch=z9hG4bK2ae9f2

CONTACT: <sip:vtspeech07.dc.voicetarget.com:2257;transport=Tcp;maddr=192.168.54.200;ms-opaque=6b1c435266dc81e2>;automata

CONTENT-LENGTH: 210

USER-AGENT: RTCC/3.0.0.0

CONTENT-TYPE: application/sdp

 

v=0

o=- 0 0 IN IP4 192.168.54.200

s=Microsoft Speech Server session

c=IN IP4 192.168.54.200

t=0 0

m=audio 13440 RTP/AVP 0 8 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendonly

a=ptime:20

 

[7] 2008/03/05 11:32:39: Set packet length to 20

[9] 2008/03/05 11:32:39: Resolve 172: tcp 192.168.54.200 2257

[9] 2008/03/05 11:32:39: SIP Tx tcp:192.168.54.200:2257:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.54.200:2257;branch=z9hG4bK2ae9f2

From: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf

To: <sip:2814610506@192.168.54.103>;tag=5140

Call-ID: f89e1cc4@pbx

CSeq: 1 INVITE

Contact: <sip:2814610506@192.168.54.103:2336;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2448

Content-Type: application/sdp

Content-Length: 232

 

v=0

o=- 35388 35388 IN IP4 192.168.54.103

s=-

c=IN IP4 192.168.54.103

t=0 0

m=audio 58934 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=recvonly

 

[7] 2008/03/05 11:32:39: f89e1cc4@pbx#5140: Media-aware pass-through mode

[6] 2008/03/05 11:32:39: Call hold from trunk

[9] 2008/03/05 11:32:39: SIP Rx tcp:192.168.54.200:2257:

ACK sip:2814610506@192.168.54.103:2336;transport=tcp SIP/2.0

FROM: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf

TO: <sip:2814610506@192.168.54.103>;tag=5140

CSEQ: 1 ACK

CALL-ID: f89e1cc4@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.54.200:2257;branch=z9hG4bKa050b1ce

CONTENT-LENGTH: 0

USER-AGENT: RTCC/3.0.0.0

 

 

[5] 2008/03/05 11:32:39: SIP port accept from 192.168.54.200:2728

[9] 2008/03/05 11:32:39: SIP Rx tcp:192.168.54.200:2728:

INVITE sip:7132234676@192.168.54.103:5060;user=phone SIP/2.0

FROM: <sip:2814610506@vtspeech07.dc.voicetarget.com:2257;user=phone>;epid=8D9F696238;tag=891bce9f3f

TO: <sip:7132234676@192.168.54.103:5060;user=phone>

CSEQ: 5 INVITE

CALL-ID: a8066fed-b09c-4d54-9b9d-bca36ec8415c

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.54.200:2728;branch=z9hG4bKa3f789a

CONTACT: <sip:vtspeech07.dc.voicetarget.com:2257;transport=Tcp;maddr=192.168.54.200;ms-opaque=6b1c435266dc81e2>;automata

CONTENT-LENGTH: 340

USER-AGENT: RTCC/3.0.0.0

CONTENT-TYPE: application/sdp

ALLOW: UPDATE

ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

 

v=0

o=- 0 0 IN IP4 192.168.54.200

s=Microsoft Speech Server session

c=IN IP4 192.168.54.200

t=0 0

m=audio 63872 RTP/AVP 114 115 4 0 8 97 101

a=rtpmap:114 x-msrta/16000

a=fmtp:114 bitrate=29000

a=rtpmap:115 x-msrta/8000

a=fmtp:115 bitrate=11800

a=rtpmap:97 RED/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

[7] 2008/03/05 11:32:39: UDP: Opening socket on port 55368

[7] 2008/03/05 11:32:39: UDP: Opening socket on port 55369

[5] 2008/03/05 11:32:39: Identify trunk (IP address and domain match) 4

[9] 2008/03/05 11:32:39: Resolve 173: tcp 192.168.54.200 2728

[9] 2008/03/05 11:32:39: SIP Tx tcp:192.168.54.200:2728:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.54.200:2728;branch=z9hG4bKa3f789a

From: <sip:2814610506@vtspeech07.dc.voicetarget.com:2257;user=phone>;epid=8D9F696238;tag=891bce9f3f

To: <sip:7132234676@192.168.54.103:5060;user=phone>;tag=f73b3fc676

Call-ID: a8066fed-b09c-4d54-9b9d-bca36ec8415c

CSeq: 5 INVITE

Content-Length: 0

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After upgrading to the latest version of PBXNSIP, we are no longer getting hold music when we do a supervised transfer. As you can see in the call log below, pbxnsip is not attempting to play moh.wav, but it does say "call hold from trunk". We haven't changed anything in our application, just upgraded to the latest version of pbxnsip. Any ideas?

 

You mean the PBX plays noise after receiving the 302 from the speech server? That sounds perfectly okay to me...? I can't imagine that a redirect would cause music on hold even in older versions.

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You mean the PBX plays noise after receiving the 302 from the speech server? That sounds perfectly okay to me...? I can't imagine that a redirect would cause music on hold even in older versions.

 

It used to always play music on hold. This is a supervised transfer, where the caller hears hold music while the server dials a new call, waits for confirmation from the called party, then connects the two parties. so it's not strictly a redirect. i was a little surprised that we were getting pbxnsip music on hold from a MSS supervised transfer, but that is how it used to work.

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It used to always play music on hold. This is a supervised transfer, where the caller hears hold music while the server dials a new call, waits for confirmation from the called party, then connects the two parties. so it's not strictly a redirect. i was a little surprised that we were getting pbxnsip music on hold from a MSS supervised transfer, but that is how it used to work.

 

Oh yes:

 

[6] 2008/03/05 11:32:39: Call hold from trunk

 

Well, there was a change in functionality. We had a customer that convinced us that it is good practice to treat call hold on a trunk differently.

 

v=0
o=- 0 0 IN IP4 192.168.54.200
s=Microsoft Speech Server session
c=IN IP4 192.168.54.200
t=0 0
m=audio 13440 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendonly
a=ptime:20

 

If I read RFC3264 correctly, the speech server says "I will only send you media", and the PBX then would just play the music back. It would be very interesting to know if the server really sends some media during that time, and the second question would be if that is actually music. Maybe there is a setting on the Speech Server that defines how to provide music on hold?

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