[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
Thu Apr 27 06:35:11 CDT 2017
[ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236745#comment-236745 ]
Andrey Zharkov commented on ASTERISK-13145:
-------------------------------------------
@Gareth:
Hello.
I have message 'WARNING' each time phone (7911, 7965, 7975) trying to register.
[Apr 25 18:46:01\] WARNING[1956\]: chan_sip.c:4293 retrans_pkt: Timeout on 002155d4-dd9b0003-e22d84cd-991ec341 at 10.10.10.113 on non-critical invite transaction.
[Apr 25 18:48:56\] WARNING[1956\]: chan_sip.c:4293 retrans_pkt: Timeout on 002155d4-dd9b0003-e22d84cd-991ec341 at 10.10.10.113 on non-critical invite transaction.
[Apr 25 18:51:51\] WARNING[1956\]: chan_sip.c:4293 retrans_pkt: Timeout on 002155d4-dd9b0003-e22d84cd-991ec341 at 10.10.10.113 on non-critical invite transaction.
[Apr 25 18:54:47\] WARNING[1956\]: chan_sip.c:4293 retrans_pkt: Timeout on 002155d4-dd9b0003-e22d84cd-991ec341 at 10.10.10.113 on non-critical invite transaction.
[Apr 25 18:57:42\] WARNING[1956\]: chan_sip.c:4293 retrans_pkt: Timeout on 002155d4-dd9b0003-e22d84cd-991ec341 at 10.10.10.113 on non-critical invite transaction.
This is sip debug of phone 7911 from my asterisk.
{noformat}
<--- SIP read from TCP:10.10.10.113:51224 --->
REGISTER sip:10.10.10.34 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.113:51224;branch=z9hG4bKb9fbf940
From: <sip:105 at 10.10.10.34>;tag=002155d4dd9b000eedd6bf64-179ff470
To: <sip:105 at 10.10.10.34>
Call-ID: 002155d4-dd9b0002-f3e190a8-e1b75a70 at 10.10.10.113
Max-Forwards: 70
Date: Tue, 25 Apr 2017 14:36:23 GMT
CSeq: 105 REGISTER
User-Agent: Cisco-CP7911G/9.4.2
Contact: <sip:105 at 10.10.10.113:51224;transport=tcp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-002155d4dd9b>";+u.sip!devicename.ccm.cisco.com="SEP002155D4DD9B";+u.sip!model.ccm.cisco.com="307"
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-6.0.0,X-cisco-xsi-8.5.1
Content-Length: 0
Expires: 180
<------------->
--- (13 headers 0 lines) ---
Sending to 10.10.10.113:51224 (no NAT)
Sending to 10.10.10.113:51224 (no NAT)
<--- Transmitting (no NAT) to 10.10.10.113:51224 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 10.10.10.113:51224;branch=z9hG4bKb9fbf940;received=10.10.10.113
From: <sip:105 at 10.10.10.34>;tag=002155d4dd9b000eedd6bf64-179ff470
To: <sip:105 at 10.10.10.34>;tag=as4551ace3
Call-ID: 002155d4-dd9b0002-f3e190a8-e1b75a70 at 10.10.10.113
CSeq: 105 REGISTER
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, REGISTER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7df4c9db"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '002155d4-dd9b0002-f3e190a8-e1b75a70 at 10.10.10.113' in 32000 ms (Method: REGISTER)
<--- SIP read from TCP:10.10.10.113:51224 --->
REGISTER sip:10.10.10.34 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.113:51224;branch=z9hG4bK5a590cf4
From: <sip:105 at 10.10.10.34>;tag=002155d4dd9b000eedd6bf64-179ff470
To: <sip:105 at 10.10.10.34>
Call-ID: 002155d4-dd9b0002-f3e190a8-e1b75a70 at 10.10.10.113
Max-Forwards: 70
Date: Tue, 25 Apr 2017 14:36:23 GMT
CSeq: 106 REGISTER
User-Agent: Cisco-CP7911G/9.4.2
Contact: <sip:105 at 10.10.10.113:51224;transport=tcp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-002155d4dd9b>";+u.sip!devicename.ccm.cisco.com="SEP002155D4DD9B";+u.sip!model.ccm.cisco.com="307"
Authorization: Digest username="105",realm="asterisk",uri="sip:10.10.10.34",response="49510769cc6cc7fa783ac159257d359d",nonce="7df4c9db",algorithm=MD5
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-6.0.0,X-cisco-xsi-8.5.1
Content-Length: 0
Expires: 180
<------------->
--- (14 headers 0 lines) ---
Sending to 10.10.10.113:51224 (no NAT)
Reliably Transmitting (no NAT) to 10.10.10.113:51224:
OPTIONS sip:105 at 10.10.10.113:51224;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.34:5060;branch=z9hG4bK7677a278
Max-Forwards: 70
From: <sip:Unknown at 10.10.10.34>;tag=as0404bd82
To: <sip:105 at 10.10.10.113:51224;transport=tcp>
Contact: <sip:Unknown at 10.10.10.34:5060;transport=TCP>
Call-ID: 746b539713b8796d0151437b36aae678 at 10.10.10.34:5060
CSeq: 101 OPTIONS
User-Agent: Asterisk PBX 11.25.1
Date: Tue, 25 Apr 2017 14:36:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, REGISTER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Content-Length: 0
---
<--- Reliably Transmitting (no NAT) to 10.10.10.113:51224 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.113:51224;branch=z9hG4bK5a590cf4;received=10.10.10.113
From: <sip:105 at 10.10.10.34>;tag=002155d4dd9b000eedd6bf64-179ff470
To: <sip:105 at 10.10.10.34>;tag=as4551ace3
Call-ID: 002155d4-dd9b0002-f3e190a8-e1b75a70 at 10.10.10.113
CSeq: 106 REGISTER
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, REGISTER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer,X-cisco-sis-6.0.0
Expires: 180
Contact: <sip:105 at 10.10.10.113:51224;transport=tcp>;expires=180
Content-Type: application/x-cisco-remotecc-response+xml
Date: Tue, 25 Apr 2017 14:36:24 GMT
Content-Length: 370
<x-cisco-remotecc-response>
<response>
<code>200</code>
<optionsind>
<combine max="6">
<remotecc><status /></remotecc>
<service-control />
</combine>
<dialog usage="hook status"><unot /></dialog>
<dialog usage="shared line"><unot /></dialog>
<presence usage="blf speed dial"><unot /></presence>
<joinreq></joinreq>
</optionsind>
</response>
</x-cisco-remotecc-response>
<------------>
Scheduling destruction of SIP dialog '1689cadc122555ea303175e6024697f2 at 10.10.10.34:5060' in 32000 ms (Method: REFER)
Reliably Transmitting (no NAT) to 10.10.10.113:51224:
REFER sip:105 at 10.10.10.113:51224;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.34:5060;branch=z9hG4bK1c409469
Max-Forwards: 70
From: <sip:Unknown at 10.10.10.34>;tag=as25394a31
To: <sip:105 at 10.10.10.113:51224;transport=tcp>
Contact: <sip:Unknown at 10.10.10.34:5060;transport=TCP>
Call-ID: 1689cadc122555ea303175e6024697f2 at 10.10.10.34:5060
CSeq: 101 REFER
User-Agent: Asterisk PBX 11.25.1
Date: Tue, 25 Apr 2017 14:36:24 GMT
Require: norefersub
Refer-To: cid:04481db7
Content-Id: 04481db7
Content-Type: multipart/mixed; boundary=uniqueBoundary
Expires: 0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, REGISTER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Content-Length: 703
--uniqueBoundary
Content-Type: application/x-cisco-remotecc-request+xml
<x-cisco-remotecc-request>
<dndupdate>
<state>disable</state>
<option>callreject</option>
</dndupdate>
</x-cisco-remotecc-request>
--uniqueBoundary
Content-Type: application/x-cisco-remotecc-request+xml
<x-cisco-remotecc-request>
<hlogupdate>
<status>on</status>
</hlogupdate>
</x-cisco-remotecc-request>
--uniqueBoundary
Content-Type: application/x-cisco-remotecc-request+xml
<x-cisco-remotecc-request>
<bulkupdate>
<contact line="1">
<mwi>no</mwi>
<cfwdallupdate>
<fwdaddress></fwdaddress>
<tovoicemail>off</tovoicemail>
</cfwdallupdate>
</contact>
</bulkupdate>
</x-cisco-remotecc-request>
--uniqueBoundary--
---
Scheduling destruction of SIP dialog '002155d4-dd9b0002-f3e190a8-e1b75a70 at 10.10.10.113' in 32000 ms (Method: REGISTER)
<--- SIP read from TCP:10.10.10.113:51224 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.34:5060;branch=z9hG4bK7677a278
From: <sip:Unknown at 10.10.10.34>;tag=as0404bd82
To: <sip:105 at 10.10.10.113:51224;transport=tcp>;tag=002155d4dd9b0010942a5eb0-399b0794
Call-ID: 746b539713b8796d0151437b36aae678 at 10.10.10.34:5060
Date: Tue, 25 Apr 2017 14:36:23 GMT
CSeq: 101 OPTIONS
Server: Cisco-CP7911G/9.4.2
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
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-6.0.0
Content-Length: 316
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 23775 0 IN IP4 10.10.10.113
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 102 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 14 lines) ---
<--- SIP read from TCP:10.10.10.113:51224 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.34:5060;branch=z9hG4bK1c409469
From: <sip:Unknown at 10.10.10.34>;tag=as25394a31
To: <sip:105 at 10.10.10.113:51224;transport=tcp>;tag=002155d4dd9b000f71610104-3576d090
Call-ID: 1689cadc122555ea303175e6024697f2 at 10.10.10.34:5060
Date: Tue, 25 Apr 2017 14:36:23 GMT
CSeq: 101 REFER
Server: Cisco-CP7911G/9.4.2
Contact: <sip:105 at 10.10.10.113:51224;transport=TCP>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '746b539713b8796d0151437b36aae678 at 10.10.10.34:5060' Method: OPTIONS
[Apr 25 18:36:56] WARNING[1956]: chan_sip.c:4231 retrans_pkt: Retransmission timeout reached on transmission 002155d4-dd9b0002-f3e190a8-e1b75a70 at 10.10.10.113 for seqno 0 (Non-critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
[Apr 25 18:36:56] WARNING[1956]: chan_sip.c:4293 retrans_pkt: Timeout on 002155d4-dd9b0002-f3e190a8-e1b75a70 at 10.10.10.113 on non-critical invite transaction.
{noformat}
What could cause this 'WARNING' messages?
Thank you.
> [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: David McNett
> Assignee: Gareth Palmer
> Attachments: 00_READ_ME_FIRST.txt, cisco-usecallmanager-11.25.1.patch, cisco-usecallmanager-13.15.0.patch, dialtemplate.xml, featurepolicy.xml, SEP000000000000.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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