[Asterisk-Users] *8 problem still there?

Wed May 19 09:28:38 MST 2004

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:luis at 192.168.2.175 ---> sip:192.168.2.176:5062)
> messenger ( sip:114 --> sip:lia at 192.168.2.175 ---> sip:192.168.2.179:16616 )
> xlite ( sip:jorge at 192.168.2.175 ---> 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 114 at asterisk **************:
> U 192.168.2.176:5062 -> 192.168.2.175:5060
>   INVITE sip:114 at 192.168.2.175 SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.176:5062;rport
> CSeq: 5406 INVITE
> To: <sip:114 at 192.168.2.175>..Content-Type: application/sdp
> From: "Luis Vazquez" <sip:luis at 192.168.2.175>;tag=E340D0A
> Call-ID: 1687598931 at 192.168.2.176
> Subject: sip:luis at 192.168.2.175
> Content-Length: 187
> User-Agent: kphone/4.0.2
> Contact: "Luis Vazquez" <sip:luis at 192.168.2.176:5062;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:luis at 192.168.2.175>;tag=E340D0A
> To: <sip:114 at 192.168.2.175>;tag=as38ce4ffc
> Call-ID: 1687598931 at 192.168.2.176
> CSeq: 5406 INVITE
> User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:114 at 192.168.2.175>
> 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:111 at ipcontact.com.uy>;tag=as3d3529c2
> To: <sip:192.168.2.179:16616>
> Contact: <sip:111 at 192.168.2.175>
> Call-ID: 4b188f24540489523fc9751f124e8068 at 192.168.2.175
> 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:luis at 192.168.2.175>;tag=E340D0A
> To:<sip:114 at 192.168.2.175>;tag=as38ce4ffc..Call-ID: 1687598931 at 192.168.2.176
> CSeq: 5406 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:114 at 192.168.2.175>
> 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:111 at ipcontact.com.uy>;tag=as3d3529c2
> To: <sip:192.168.2.179:16616>;tag=b271370b-aeed-4640-adca-d60c86b188d7
> Call-ID: 4b188f24540489523fc9751f124e8068 at 192.168.2.175
> 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:111 at ipcontact.com.uy>;tag=as3d3529c2
> To: <sip:192.168.2.179:16616>;tag=b271370b-aeed-4640-adca-d60c86b188d7
> Call-ID: 4b188f24540489523fc9751f124e8068 at 192.168.2.175
> CSeq: 102 INVITE
> User-Agent: Windows RTC/1.0
> Content-Length: 0
> 
> #
> ********* Here starts call pickup ***************
> 
> *********** Xlite enters the game sending an Invite to *8 at asterisk *******:
> U 192.168.2.179:5061 -> 192.168.2.175:5060
>   INVITE sip:*8 at ipcontact.com.uy SIP/2.0
> Via: SIP/2.0/UDP 
> 192.168.2.179:5061;rport;branch=z9hG4bK41344F5B08434749A915F0DCFAB2AE66
> From: Jorge <sip:jorge at ipcontact.com.uy:5061>;tag=1940958518
> To: <sip:*8 at ipcontact.com.uy>
> Contact: <sip:jorge at 192.168.2.179:5061>
> Call-ID:BE331F89-45C6-48DA-85F8-F3AD91A827A7 at 192.168.2.179
> 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:jorge at ipcontact.com.uy:5061>;tag=1940958518
> To: <sip:*8 at ipcontact.com.uy>;tag=as4b041d55
> Call-ID: BE331F89-45C6-48DA-85F8-F3AD91A827A7 at 192.168.2.179
> CSeq: 19484 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:*8 at 192.168.2.175>
> 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:jorge at ipcontact.com.uy:5061>;tag=1940958518
> To: <sip:*8 at ipcontact.com.uy>;tag=as4b041d55
> Call-ID:BE331F89-45C6-48DA-85F8-F3AD91A827A7 at 192.168.2.179
> CSeq: 19484 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:*8 at 192.168.2.175>
> 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:luis at 192.168.2.175>;tag=E340D0A
> To: <sip:114 at 192.168.2.175>;tag=as38ce4ffc
> Call-ID: 1687598931 at 192.168.2.176
> CSeq: 5406 INVITE..User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:114 at 192.168.2.175>..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:114 at 192.168.2.175 SIP/2.0..Via: SIP/2.0/UDP 
> 192.168.2.176:5062;rport
> CSeq: 5406 ACK
> To: <sip:114 at 192.168.2.175>;tag=as38ce4ffc
> From: "Luis Vazquez" <sip:luis at 192.168.2.175>;tag=E340D0A
> Call-ID:1687598931 at 192.168.2.176
> Content-Length: 0..User-Agent: kphone/4.0.2
> Contact: "Luis Vazquez" <sip:luis at 192.168.2.176:5062;transport=udp>
> 
> #
> U 192.168.2.179:5061 -> 192.168.2.175:5060
>   ACK sip:*8 at 192.168.2.175 SIP/2.0..Via: SIP/2.0/UDP 
> 192.168.2.179:5061;rport;branch=z9hG4bK4CE63D00AB944D4CB7BED0D3A2B8B939
> From: Jorge <sip:jorge at ipcontact.com.uy:5061>;tag=1940958518
> To: <sip:*8 at ipcontact.com.uy>;tag=as4b041d55
> Contact: <sip:jorge at 192.168.2.179:5061>
> Call-ID: BE331F89-45C6-48DA-85F8-F3AD91A827A7 at 192.168.2.179
> CSeq: 19484 ACK..Max-Forwards: 70
> Content-Length: 0
> 
> 
> What happened with our friend lia at messenger????
> 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
> 
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