Thanks to all for the comments even if they don't agree! I think this issue is significant and I would really like it to be fixed in the 1.0 release.
Does anybody know how to get the same functionality without using *8? -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Vazquez Sent: Wednesday, May 19, 2004 8:48 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] *8 problem still there? 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=z9hG4bK41344F5B08434749A915F0DCFA B2AE66 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
