FYI I see it only on 1 in about 10-20 pickups... On Wed, 19 May 2004, Luis Vazquez wrote:
> Shaun Ewing wrote: > > >I'm not seeing this - using stable CVS from 14-05-2004. > > > >Phone types are Grandstream Budgetone 101 using firmware 1.0.4.68, and Cisco > >7940 using SIP 6.2. > > > >-Shaun > > > > > > > Just to give more info. > I just made a testing using stable CVS from 24-04-2004 and 3 softphone > clients registered in asterisk with users > luis, lia and jorge (with fromdomain=ipcontact.com.uy in sip.conf): > > kphone ( sip:111 ---> sip:[EMAIL PROTECTED] ---> sip:192.168.2.176:5062) > messenger ( sip:114 --> sip:[EMAIL PROTECTED] ---> sip:192.168.2.179:16616 ) > xlite ( sip:[EMAIL PROTECTED] ---> sip:192.168.2.179:5061) > > Here is the dialog in a call from luis(kphone) to 114(messenger) and a > pickup with *8 from jorge(xlite). > > The kphone and xlite get connected but 114 (lia - messenger) never gets > a CANCEL: > > ****Invite from luis to [EMAIL PROTECTED] **************: > U 192.168.2.176:5062 -> 192.168.2.175:5060 > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.176:5062;rport > CSeq: 5406 INVITE > To: <sip:[EMAIL PROTECTED]>..Content-Type: application/sdp > From: "Luis Vazquez" <sip:[EMAIL PROTECTED]>;tag=E340D0A > Call-ID: [EMAIL PROTECTED] > Subject: sip:[EMAIL PROTECTED] > Content-Length: 187 > User-Agent: kphone/4.0.2 > Contact: "Luis Vazquez" <sip:[EMAIL PROTECTED]:5062;transport=udp> > > v=0..o=username 0 0 IN IP4 192.168.2.176..s=The Funky Flow > c=IN IP4 192.168.2.176..t=0 0 > m=audio 32842 RTP/AVP 0 97 3 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000.. > # > U 192.168.2.175:5060 -> 192.168.2.176:5062 > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.2.176:5062;rport;received=192.168.2.176 > From: "Luis Vazquez" <sip:[EMAIL PROTECTED]>;tag=E340D0A > To: <sip:[EMAIL PROTECTED]>;tag=as38ce4ffc > Call-ID: [EMAIL PROTECTED] > CSeq: 5406 INVITE > User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0.... > # > > ********** Relay of Invite from asterisk to messenger***************: > U 192.168.2.175:5060 -> 192.168.2.179:16616 > INVITE sip:192.168.2.179:16616 SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.175:5060;branch=z9hG4bK535d9d2c > From: "Luis(1084976431.475)" <sip:[EMAIL PROTECTED]>;tag=as3d3529c2 > To: <sip:192.168.2.179:16616> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX..Date: Wed, 19 May 2004 14:20:33 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > UniqueID: 1084976433.476 > Content-Type: application/sdp > Content-Length: 211 > > v=0 > o=root 20766 20766 IN IP4 192.168.2.175 > s=session > c=IN IP4 192.168.2.175..t=0 0 > m=audio 17996 RTP/AVP 0 397 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=silenceSupp:off - - - -.. > # > > *********** Asterisk says to kphone messenger is ringing **************: > U 192.168.2.175:5060 -> 192.168.2.176:5062 > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.2.176:5062;rport;received=192.168.2.176 > From: "Luis Vazquez" <sip:[EMAIL PROTECTED]>;tag=E340D0A > To:<sip:[EMAIL PROTECTED]>;tag=as38ce4ffc..Call-ID: [EMAIL PROTECTED] > CSeq: 5406 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > # > > ********** Messenger says to Asterisk he is trying ******************: > U 192.168.2.179:1071 -> 192.168.2.175:5060 > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.2.175:5060;branch=z9hG4bK535d9d2c > From: "Luis(1084976431.475)" <sip:[EMAIL PROTECTED]>;tag=as3d3529c2 > To: <sip:192.168.2.179:16616>;tag=b271370b-aeed-4640-adca-d60c86b188d7 > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Windows RTC/1.0 > Content-Length: 0 > > # > > *********** Messenger is ringing (and will be forever if not anwered) > ****************: > U 192.168.2.179:1071 -> 192.168.2.175:5060 > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.2.175:5060;branch=z9hG4bK535d9d2c > From: "Luis(1084976431.475)" <sip:[EMAIL PROTECTED]>;tag=as3d3529c2 > To: <sip:192.168.2.179:16616>;tag=b271370b-aeed-4640-adca-d60c86b188d7 > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Windows RTC/1.0 > Content-Length: 0 > > # > ********* Here starts call pickup *************** > > *********** Xlite enters the game sending an Invite to [EMAIL PROTECTED] *******: > U 192.168.2.179:5061 -> 192.168.2.175:5060 > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP > 192.168.2.179:5061;rport;branch=z9hG4bK41344F5B08434749A915F0DCFAB2AE66 > From: Jorge <sip:[EMAIL PROTECTED]:5061>;tag=1940958518 > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]:5061> > Call-ID:[EMAIL PROTECTED] > CSeq: 19484 INVITE > Max-Forwards: 70 > Content-Type: application/sdp > User-Agent: X-Lite release 1103a > Content-Length: 193 > > v=0..o=jorge 2391140 2391203 IN IP4 192.168.2.179 > s=X-Lite > c=IN IP4 192.168.2.179..t=0 0 > m=audio 8000 RTP/AVP 3 101 > a=rtpmap:3 gsm/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15.. > # > > ******** Asterisk responds he is trying **************: > U 192.168.2.175:5060 -> 192.168.2.179:5061 > SIP/2.0 100 Trying > Via: > SIP/2.0/UDP192.168.2.179:5061;rport;branch=z9hG4bK41344F5B08434749A915F0DCFAB2AE66 > From: Jorge <sip:[EMAIL PROTECTED]:5061>;tag=1940958518 > To: <sip:[EMAIL PROTECTED]>;tag=as4b041d55 > Call-ID: [EMAIL PROTECTED] > CSeq: 19484 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > > # > > ********* Asterisk accept the call from the Xlite (jorge) *****: > U 192.168.2.175:5060 -> 192.168.2.179:5061 > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.2.179:5061;rport;branch=z9hG4bK41344F5B08434749A915F0DCFAB2AE66 > From: Jorge <sip:[EMAIL PROTECTED]:5061>;tag=1940958518 > To: <sip:[EMAIL PROTECTED]>;tag=as4b041d55 > Call-ID:[EMAIL PROTECTED] > CSeq: 19484 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > UniqueID:1084976436.477 > Content-Type: application/sdp > Content-Length: 215 > > v=0 > o=root 3611 3611 IN IP4 192.168.2.175 > s=session > c=IN IP4 192.168.2.175 > t=0 0 > m=audio 16578 RTP/AVP 3 101 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - -.. > # > > ****** Asterisk accept the call from kphone and bridges with xlite ******: > U 192.168.2.175:5060 -> 192.168.2.176:5062 > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.2.176:5062;rport;received=192.168.2.176 > From: "Luis Vazquez"<sip:[EMAIL PROTECTED]>;tag=E340D0A > To: <sip:[EMAIL PROTECTED]>;tag=as38ce4ffc > Call-ID: [EMAIL PROTECTED] > CSeq: 5406 INVITE..User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]>..UniqueID: 1084976431.475 > Content-Type: application/sdp > Content-Length: 162 > > v=0 > o=root 20766 20766 IN IP4 192.168.2.175 > s=session > c=IN IP4 192.168.2.175 > t=0 0 > m=audio 16964 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > a=silenceSupp:off > > # > > ***************** Both clients sed theirs ACKs and get connected > **************: > U 192.168.2.176:5062 -> 192.168.2.175:5060 > ACK sip:[EMAIL PROTECTED] SIP/2.0..Via: SIP/2.0/UDP > 192.168.2.176:5062;rport > CSeq: 5406 ACK > To: <sip:[EMAIL PROTECTED]>;tag=as38ce4ffc > From: "Luis Vazquez" <sip:[EMAIL PROTECTED]>;tag=E340D0A > Call-ID:[EMAIL PROTECTED] > Content-Length: 0..User-Agent: kphone/4.0.2 > Contact: "Luis Vazquez" <sip:[EMAIL PROTECTED]:5062;transport=udp> > > # > U 192.168.2.179:5061 -> 192.168.2.175:5060 > ACK sip:[EMAIL PROTECTED] SIP/2.0..Via: SIP/2.0/UDP > 192.168.2.179:5061;rport;branch=z9hG4bK4CE63D00AB944D4CB7BED0D3A2B8B939 > From: Jorge <sip:[EMAIL PROTECTED]:5061>;tag=1940958518 > To: <sip:[EMAIL PROTECTED]>;tag=as4b041d55 > Contact: <sip:[EMAIL PROTECTED]:5061> > Call-ID: [EMAIL PROTECTED] > CSeq: 19484 ACK..Max-Forwards: 70 > Content-Length: 0 > > > What happened with our friend [EMAIL PROTECTED] > She is still ringing and waiting for a CANCEL a BYE or something. > And that's all. > > Just in case here is the sip.conf > [general] > port = 5060 > context = local > .......... > [luis] > type = friend > callgroup=2 > pickupgroup=2 > username = luis > host = dynamic > disallow=all > allow=ulaw > allow=gsm > dtmfmode=inband > callerid="Luis" <111> > > [jorge] > type = friend > callgroup=2 > pickupgroup=2 > username = jorge > disallow=all > allow=gsm > dtmfmode=rfc2833 > host = dynamic > callerid="Jorge" <112> > > [lia] > type = friend > callgroup=2 > pickupgroup=2 > username = lia > dtmfmode=inband > host = dynamic > callerid="Lia" <114> > > I hope someone have the time and patience to take a look. > Godbye > Luis > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
