[asterisk-bugs] [JIRA] (ASTERISK-13145) [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML
Andrey Zharkov (JIRA)
noreply at issues.asterisk.org
Mon Feb 4 04:10:58 CST 2019
[ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=246126#comment-246126 ]
Andrey Zharkov commented on ASTERISK-13145:
-------------------------------------------
@ Gareth.
I changed the callerIdBlocking to 0.
Now calls from the second line do not pass at all.
sip debug 108 (second line) -> 106
{noformat}
<--- SIP read from TCP:10.10.11.117:51498 --->
INVITE sip:100 at 10.10.10.34;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.11.117:51498;branch=z9hG4bK1fdf86f2
From: "108" <sip:108 at 10.10.10.34>;tag=00b1e3bb7fbe001c0526129b-3eb3347e
To: <sip:100 at 10.10.10.34>
Call-ID: 00b1e3bb-7fbe000e-12540488-7ab51bc0 at 10.10.11.117
Max-Forwards: 70
Session-ID: 5288974300105000a00000b1e3bb7fbe;remote=00000000000000000000000000000000
Date: Mon, 04 Feb 2019 09:51:53 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP8865/12.1.1
Contact: <sip:108 at 10.10.11.117:51498;user=phone;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00B1E3BB7FBE";video
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "108" <sip:108 at 10.10.10.34>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="107",realm="asterisk",uri="sip:100 at 10.10.10.34;user=phone",response="df4c2025f9390a63d66e4cd0d0cfd11e",nonce="1272a416",algorithm=MD5
Content-Length: 1183
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 10924 0 IN IP4 10.10.11.117
s=SIP Call
b=AS:4064
t=0 0
m=audio 19546 RTP/AVP 0 8 116 18 101
c=IN IP4 10.10.11.117
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 19778 RTP/AVP 100 126 97
c=IN IP4 10.10.11.117
b=TIAS:4000000
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=640C16;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428016;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
<------------->
--- (23 headers 33 lines) ---
Sending to 10.10.11.117:51498 (no NAT)
Sending to 10.10.11.117:51498 (no NAT)
Using INVITE request as basis request - 00b1e3bb-7fbe000e-12540488-7ab51bc0 at 10.10.11.117
Found peer '108' for '108' from 10.10.11.117:51498
<--- Reliably Transmitting (no NAT) to 10.10.11.117:51498 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 10.10.11.117:51498;branch=z9hG4bK1fdf86f2;received=10.10.11.117
From: "108" <sip:108 at 10.10.10.34>;tag=00b1e3bb7fbe001c0526129b-3eb3347e
To: <sip:100 at 10.10.10.34>;tag=as070d65f6
Call-ID: 00b1e3bb-7fbe000e-12540488-7ab51bc0 at 10.10.11.117
CSeq: 101 INVITE
Server: Asterisk PBX 13.24.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f05bd6b"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '00b1e3bb-7fbe000e-12540488-7ab51bc0 at 10.10.11.117' in 32000 ms (Method: INVITE)
<--- SIP read from TCP:10.10.11.117:51498 --->
ACK sip:100 at 10.10.10.34;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.11.117:51498;branch=z9hG4bK1fdf86f2
From: "108" <sip:108 at 10.10.10.34>;tag=00b1e3bb7fbe001c0526129b-3eb3347e
To: <sip:100 at 10.10.10.34>;tag=as070d65f6
Call-ID: 00b1e3bb-7fbe000e-12540488-7ab51bc0 at 10.10.11.117
Session-ID: 5288974300105000a00000b1e3bb7fbe;remote=00000000000000000000000000000000
Max-Forwards: 70
Date: Mon, 04 Feb 2019 09:51:53 GMT
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from TCP:10.10.11.117:51498 --->
INVITE sip:100 at 10.10.10.34;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.11.117:51498;branch=z9hG4bK5be51999
From: "108" <sip:108 at 10.10.10.34>;tag=00b1e3bb7fbe001c0526129b-3eb3347e
To: <sip:100 at 10.10.10.34>
Call-ID: 00b1e3bb-7fbe000e-12540488-7ab51bc0 at 10.10.11.117
Max-Forwards: 70
Session-ID: 5288974300105000a00000b1e3bb7fbe;remote=00000000000000000000000000000000
Date: Mon, 04 Feb 2019 09:51:53 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP8865/12.1.1
Contact: <sip:108 at 10.10.11.117:51498;user=phone;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00B1E3BB7FBE";video
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "108" <sip:108 at 10.10.10.34>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="107",realm="asterisk",uri="sip:100 at 10.10.10.34;user=phone",response="a51c25c58d3dfcde72c3683abfb8c000",nonce="6f05bd6b",algorithm=MD5
Content-Length: 1183
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 10924 0 IN IP4 10.10.11.117
s=SIP Call
b=AS:4064
t=0 0
m=audio 19546 RTP/AVP 0 8 116 18 101
c=IN IP4 10.10.11.117
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 19778 RTP/AVP 100 126 97
c=IN IP4 10.10.11.117
b=TIAS:4000000
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=640C16;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428016;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
<------------->
--- (23 headers 33 lines) ---
Sending to 10.10.11.117:51498 (no NAT)
Using INVITE request as basis request - 00b1e3bb-7fbe000e-12540488-7ab51bc0 at 10.10.11.117
Found peer '108' for '108' from 10.10.11.117:51498
[Feb 2 20:47:06] WARNING[3025][C-0000000e]: chan_sip.c:17964 check_auth: username mismatch, have <108>, digest has <107>
[Feb 2 20:47:06] NOTICE[3025][C-0000000e]: chan_sip.c:30090 handle_request_invite: Failed to authenticate device "108" <sip:108 at 10.10.10.34>;tag=00b1e3bb7fbe001c0526129b-3eb3347e
<--- Reliably Transmitting (no NAT) to 10.10.11.117:51498 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/TCP 10.10.11.117:51498;branch=z9hG4bK5be51999;received=10.10.11.117
From: "108" <sip:108 at 10.10.10.34>;tag=00b1e3bb7fbe001c0526129b-3eb3347e
To: <sip:100 at 10.10.10.34>;tag=as070d65f6
Call-ID: 00b1e3bb-7fbe000e-12540488-7ab51bc0 at 10.10.11.117
CSeq: 102 INVITE
Server: Asterisk PBX 13.24.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Call-Info: <urn:x-cisco-remotecc:callinfo>; security=NotAuthenticated
Content-Length: 0
<------------>
{noformat}
> [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML
> -------------------------------------------------------------------------------
>
> Key: ASTERISK-13145
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-13145
> Project: Asterisk
> Issue Type: New Feature
> Components: Channels/chan_sip/NewFeature
> Reporter: Gareth Palmer
> Assignee: Gareth Palmer
> Labels: patch, pjsip
> Attachments: 00_READ_ME_FIRST.txt, AppDialRules.xml, cisco-usecallmanager-13.24.1.patch, cisco-usecallmanager-16.1.1.patch, DialTemplate.xml, FeaturePolicy.xml, SEPMAC.cnf.xml, SoftKeys.xml
>
>
> This patch provides support for Cisco 6900, 7900, 8800 and 9900 series phones using the SIP firmware.
> Available features are: Busy Lamp Field, Off Hook Notification, Call Forward, Do Not Disturb, Huntgroup Login, Call Park (Notify and Monitor), Server-Side Ad-Hoc Conference, Conference List, Kick and Mute/Unmute, Multi-Admin Conference, Multiple Lines via Bulk Register, Immediate Divert, Call Recording, Restart or Reset via CLI, Call Pickup Notification, Call Back, Join Calls, Mallicious Call ID, Quality Reporting Tool and Fail-over/Fail-back.
> Also included is Application Server Events used by non-USECALLMANAGER phones (Call Forward and Do Not Disturb only).
> *Important:* Read the documentation at [http://usecallmanager.nz] to see the additional configuration options required for the phones to operate correctly.
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