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No audible ring on working SIP trunk

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T

telecomtekperson

Vendor
Joined
Jan 17, 2005
Messages
1,477
Location
US
406, 2.4.11 Sip through Global Crossing.

SIp trunk works both inbound and outbound. DTMF works inbound and outbound. If you place a call to the SIP trunk, the call rings the IPO, but you never hear the ringing on the phone that dialied the SIP line.

Ever heard of that? We are sending the 180's back to them to tell them to make it ring....so I am guessing that the problem is not with the IPO.

 
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Is your Codec still on automatic?
If so try to change it to G.711 or G.729.
Or try to tick enable NAT in System > Lan1 or Lan2 tab.


Greetzzz...Bas

y1pzZTEUdok1vrI5cLb3FdPX4PgTPlSONkb5WPjz0x50etSujaMSmhdRCbOx9vASnrRNzzXv0IxNQA

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
********** contact made with 192.168.5.2 at 19:03:11 11/12/2008 **********

23511844mS SIP Rx: UDP 67.16.111.109:5060 ->MY WAN IP:5060

INVITE sip:2629231118;npdi=yes@MyWanIP:5060 SIP/2.0

Via: SIP/2.0/UDP 67.16.111.109:5060;branch=z9hG4bK0eB89337556b04c1523

From: <sip:2623310091@67.16.111.109;isup-oli=61;pstn-params=808281808882>;tag=gK0e538b13

To: <sip:2629231118@MyWanIP>

Call-ID: 1494142844_15681@67.16.111.109

CSeq: 9117 INVITE

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS

Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed

Contact: <sip:2623310091@67.16.111.109:5060>

Remote-Party-ID: <sip:2623310091@67.16.111.109:5060>;privacy=off

P-Charge-Info: sip:2623310091@67.16.111.109:5060

Supported: timer,100rel

Session-Expires: 3600

Min-SE: 90

Content-Length: 283

Content-Disposition: session; handling=required

Content-Type: application/sdp



v=0

o=Sonus_UAC 9364 2509 IN IP4 67.16.111.109

s=SIP Media Capabilities

c=IN IP4 67.16.111.110

t=0 0

m=audio 18850 RTP/AVP 0 18 100

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-15

a=sendrecv

a=maxptime:20

23511845mS SIP Trunk: 49:Rx

INVITE sip:2629231118;npdi=yes@MyWanIP:5060 SIP/2.0

Via: SIP/2.0/UDP 67.16.111.109:5060;branch=z9hG4bK0eB89337556b04c1523

From: <sip:2623310091@67.16.111.109;isup-oli=61;pstn-params=808281808882>;tag=gK0e538b13

To: <sip:2629231118@MyWanIP>

Call-ID: 1494142844_15681@67.16.111.109

CSeq: 9117 INVITE

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS

Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed

Contact: <sip:2623310091@67.16.111.109:5060>

Remote-Party-ID: <sip:2623310091@67.16.111.109:5060>;privacy=off

P-Charge-Info: sip:2623310091@67.16.111.109:5060

Supported: timer,100rel

Session-Expires: 3600

Min-SE: 90

Content-Length: 283

Content-Disposition: session; handling=required

Content-Type: application/sdp



v=0

o=Sonus_UAC 9364 2509 IN IP4 67.16.111.109

s=SIP Media Capabilities

c=IN IP4 67.16.111.110

t=0 0

m=audio 18850 RTP/AVP 0 18 100

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-15

a=sendrecv

a=maxptime:20

23511851mS SipDebugInfo: SIPDialog TXN : Decoding of message Succeded 1

23511852mS SipDebugInfo: Create Incoming EndPoint voip

23511852mS SipDebugInfo: License, Valid 1, Available 1, Consumed 0

23511853mS SipDebugInfo: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!

23511854mS SipDebugInfo: 49.1307.1 -1 SIPTrunk Endpoint(fea6bd64) ExtractCallIdFrom Message: Call Id value 1494142844_15681@67.16.111.109

23511854mS SipDebugInfo: 49.1307.1 -1 SIPTrunk Endpoint(fea6bd64) ExtractCallerFromMessage: From Tag is gK0e538b13

23511855mS SipDebugInfo: 49.1307.1 -1 SIPTrunk Endpoint(fea6bd64) Accept Media Range

23511856mS SipDebugInfo: 49.1307.1 -1 SIPTrunk Endpoint(fea6bd64) SendSIPResponse: INVITE SENT TO 67.16.111.109 5060

23511857mS SipDebugInfo: 49.1307.1 -1 SIPTrunk Endpoint(fea6bd64) Sending code 100 to method INVITE

23511857mS SipDebugInfo: 49.1307.1 -1 SIPTrunk Endpoint(fea6bd64) SendSIPResponse, Number of Tag Count, 0



********** Warning: Missed 13 packet(s) **********

23511883mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6c988) received CMAlerting

23511883mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) SIPDialog::ExtractFastStartupElement There are 3 Information Element

23511884mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) SIPEndPoint: Received an CMAlerting State Transition to SIPDialog::INVITE_RCVD(10)

23511884mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) SendSIPResponse: INVITE SENT TO 67.16.111.109 5060

23511884mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) Sending code 180 to method INVITE

23511885mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) SendSIPResponse, Number of Tag Count, 1

23511886mS SIP Trunk: 49:Tx

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 67.16.111.109:5060;branch=z9hG4bK0eB89337556b04c1523

From: <sip:2623310091@67.16.111.109;isup-oli=61;pstn-params=808281808882>;tag=gK0e538b13

To: <sip:2629231118@MyWanIP>;tag=472ee6ebdb107bba

Call-ID: 1494142844_15681@67.16.111.109

CSeq: 9117 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO

Content-Length: 0



23511886mS SIP Tx: UDP MyWanIP:5060 -> 67.16.111.109:5060

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 67.16.111.109:5060;branch=z9hG4bK0eB89337556b04c1523

From: <sip:2623310091@67.16.111.109;isup-oli=61;pstn-params=808281808882>;tag=gK0e538b13

To: <sip:2629231118@MyWanIP>;tag=472ee6ebdb107bba

Call-ID: 1494142844_15681@67.16.111.109

CSeq: 9117 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO

Content-Length: 0



23511888mS SipDebugInfo: SIP Line (49): Freed Txn Key 2016

23520633mS SIP Rx: UDP 67.16.111.109:5060 -> MyWanIP:5060

CANCEL sip:2629231118;npdi=yes@MyWanIP:5060 SIP/2.0

Via: SIP/2.0/UDP 67.16.111.109:5060;branch=z9hG4bK0eB89337556b04c1523

From: <sip:2623310091@67.16.111.109;isup-oli=61;pstn-params=808281808882>;tag=gK0e538b13

To: <sip:2629231118@69.129.89.252>

Call-ID: 1494142844_15681@67.16.111.109

CSeq: 9117 CANCEL

Max-Forwards: 70

Content-Length: 0



23520633mS SIP Trunk: 49:Rx

CANCEL sip:2629231118;npdi=yes@MyWanIP:5060 SIP/2.0

Via: SIP/2.0/UDP 67.16.111.109:5060;branch=z9hG4bK0eB89337556b04c1523

From: <sip:2623310091@67.16.111.109;isup-oli=61;pstn-params=808281808882>;tag=gK0e538b13

To: <sip:2629231118@MyWanIP>

Call-ID: 1494142844_15681@67.16.111.109

CSeq: 9117 CANCEL

Max-Forwards: 70

Content-Length: 0



23520635mS SipDebugInfo: SIPDialog TXN : Decoding of message Succeded 1

23520636mS SipDebugInfo: Find End Point 49.1307.1 61 SIPTrunk Endpoint (fea6c988) Sip CallId 1494142844_15681@67.16.111.109

23520637mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) Process SIP request dialog fea6bd64, method CANCEL in state SIPDialog::INVITE_RCVD(10)

23520637mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) SendSIPResponse: CANCEL SENT TO 67.16.111.109 5060

23520638mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) Sending code 200 to method CANCEL

23520638mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) SendSIPResponse, Number of Tag Count, 0

23520639mS SIP Trunk: 49:Tx

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 67.16.111.109:5060;branch=z9hG4bK0eB89337556b04c1523

From: <sip:2623310091@67.16.111.109;isup-oli=61;pstn-params=808281808882>;tag=gK0e538b13

To: <sip:2629231118@MyWanIP>;tag=472ee6ebdb107bba

Call-ID: 1494142844_15681@67.16.111.109

CSeq: 9117 CANCEL

Content-Length: 0



23520639mS SIP Tx: UDP MyWanIP

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 67.16.111.109:5060;branch=z9hG4bK0eB89337556b04c1523

From: <sip:2623310091@67.16.111.109;isup-oli=61;pstn-params=808281808882>;tag=gK0e538b13

To: <sip:2629231118@MyWanIP>;tag=472ee6ebdb107bba

Call-ID: 1494142844_15681@67.16.111.109

CSeq: 9117 CANCEL

Content-Length: 0



23520640mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) SendSIPResponse: INVITE SENT TO 67.16.111.109 5060

23520640mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) Sending code 487 to method INVITE

23520641mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) SendSIPResponse, Number of Tag Count, 1

23520642mS SIP Trunk: 49:Tx

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP 67.16.111.109:5060;branch=z9hG4bK0eB89337556b04c1523

From: <sip:2623310091@67.16.111.109;isup-oli=61;pstn-params=808281808882>;tag=gK0e538b13

To: <sip:2629231118@6MyWanIP\ >;tag=472ee6ebdb107bba

Call-ID: 1494142844_15681@67.16.111.109

CSeq: 9117 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO

Content-Length: 0



23520642mS SIP Tx: UDP MyWanIP:5060 -> 67.16.111.109:5060

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP 67.16.111.109:5060;branch=z9hG4bK0eB89337556b04c1523

From: <sip:2623310091@67.16.111.109;isup-oli=61;pstn-params=808281808882>;tag=gK0e538b13

To: <sip:2629231118@MyWanIP>;tag=472ee6ebdb107bba

Call-ID: 1494142844_15681@67.16.111.109

CSeq: 9117 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO

Content-Length: 0



23520644mS SipDebugInfo: 49.1307.1 61 SIPTrunk Endpoint(fea6bd64) UpdateSIPCallState SIPDialog::INVITE_RCVD(10) -> SIPDialog::FINAL(40)

23520648mS SipDebugInfo: 49.1307.1 -1 SIPTrunk Endpoint(fea6bd64) Terminating dialog fea6bd64, state SIPDialog::FINAL(40) for cause 16

23520656mS SipDebugInfo: SIP Line (49): Freed Txn Key 2016

23520740mS SIP Rx: UDP 67.16.111.109:5060 -> 69.129.89.252:5060

ACK sip:2629231118;npdi=yes@MyWanIP:5060 SIP/2.0

Via: SIP/2.0/UDP 67.16.111.109:5060;branch=z9hG4bK0eB89337556b04c1523

From: <sip:2623310091@67.16.111.109;isup-oli=61;pstn-params=808281808882>;tag=gK0e538b13

To: <sip:2629231118@MyWanIp>;tag=472ee6ebdb107bba

Call-ID: 1494142844_15681@67.16.111.109

CSeq: 9117 ACK

Max-Forwards: 70

Content-Length: 0



23520741mS SIP Trunk: 49:Rx

ACK sip:2629231118;npdi=yes@MyWanIp:5060 SIP/2.0

Via: SIP/2.0/UDP 67.16.111.109:5060;branch=z9hG4bK0eB89337556b04c1523

From: <sip:2623310091@67.16.111.109;isup-oli=61;pstn-params=808281808882>;tag=gK0e538b13

To: <sip:2629231118@MyWanIp>;tag=472ee6ebdb107bba

Call-ID: 1494142844_15681@67.16.111.109

CSeq: 9117 ACK

Max-Forwards: 70

Content-Length: 0



23520743mS SipDebugInfo: SIPDialog TXN : Decoding of message Succeded 1

23520743mS SipDebugInfo: Find End Point 49.1307.1 -1 SIPTrunk Endpoint (fea6c988) Sip CallId 1494142844_15681@67.16.111.109

23520744mS SipDebugInfo: 49.1307.1 -1 SIPTrunk Endpoint(fea6bd64) Process SIP request dialog fea6bd64, method ACK in state SIPDialog::FINAL(40)

23521746mS SipDebugInfo: 49.1307.1 -1 SIPTrunk Endpoint(fea6bd64) EPTerminationTimeout, about to delete endpoint

23521747mS SipDebugInfo: 49.1307.1 -1 SIPTrunk Endpoint(fea6bd64) SIPDialog destructor ... fea6bd64

23521747mS SipDebugInfo: ~SipTrunkEndpoint 49.1307.1 -1 SIPTrunk Endpoint

23525641mS SipDebugInfo: Timer 10 callback

23525744mS SipDebugInfo: Timer 9 callback





 
Upvote 0 Downvote
This could be a provider problem, is i'm not mistaken the IPO is not able to produce it's own Ringback tone but the provider needs to do it, maybe other on tek-tips can confirm it.

Greetzzz...Bas

y1pzZTEUdok1vrI5cLb3FdPX4PgTPlSONkb5WPjz0x50etSujaMSmhdRCbOx9vASnrRNzzXv0IxNQA

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
The provider is sending you an invite but the dtmf mode is not rfc2833. Get this fixed first then try again with new post.
 
Upvote 0 Downvote
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