07-20-2016 02:50 PM - edited 03-17-2019 07:36 AM
I have several 9971 phones running SIP and working well on a CME 8.6 system. The CME also has plenty of SCCP phones running on it. I recently tried to add a 9971 phone that connects to the CME via VPN so therefore it is coming from a non-local subnet.
There is no firewall and no NAT involved. The phone registers but cannot call to any extension.
Doing a debug I basically see this: INVITE from 9971 require: sdp-anat from CME 420 bad extension unsupported: sdp-anat Why does the phone think it needs NAT? How do I tell the phone to not require this, or conversely, enable it on the CME to make the phone happy?
Thanks, Diego
Solved! Go to Solution.
08-04-2016 12:06 PM
Great to see this resolved.
I remember posting earlier that config file probably cached because ANAT for 9971 isn't supported by CME and probably the phone was registered with CUCM earlier which has ANAT enabled.
Regarding shared DNs for SIP and SCCP, yes you need to create voice register dn and ephone-dn with same number. If you want to have both ringing at the same time, you need to put the command shared sip on ephone-dn and shared max # on voice register dn.
Note: Shared DNs between SIP and SCCP started from CME 9.0 and its bit flaky. Some times you need to reboot the box to get it working.
07-20-2016 10:46 PM
Hi,
ANAT is required for Dual mode support (IPv4/IPv6). Just make sure the preference configured in sip-ua is IPv4 only.
07-21-2016 05:19 AM
Mohammed,
That is a good point but I don't think it applies to my situation. I don't have IPV6 enabled anywhere in my environment. Also, I believe the sip-ua command would be implemented at the CME but it is the phone that is requesting the feature. How would I disable the phone from requiring sdp-anat?
Thanks,
Diego
07-21-2016 09:21 AM
I understand by the CME needs to be enabled for ANAT to accept the message from the phone. If the phone has ANAT in supported or required headers while CME isn't enabled for ANAT, CME will send 420 message. Just enable ANAT in your CME with preference mode IPv4 and test.
sip-ua
protocol mode dual-stack preference ipv4
07-22-2016 02:10 PM
Hello Mohammed,
I added the commands to the sip-ua and reloaded the CME for good measure. It seems like IPV4/6 is now enabled but the phone is still getting the "420 bad extension" and "unsupported:sdp-anat" from the CME. Here is what the sip-ua looks like:
UC520#sho sip-ua status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): ENABLED 192.168.1.100
SIP User Agent bind status(media): ENABLED 192.168.1.100
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
protocol mode is dual-stack, preference is ipv4
SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported: audio video image
Network types supported: IN
Address types supported: IP4 IP6
Transport types supported: RTP/AVP udptl
07-23-2016 12:48 AM
Ok. Can you please share the following.
sh run | sec voice register
debug ccsip messs ---- for a test call.
07-23-2016 10:04 AM
Here you are sir. Sorry about the formatting of the debug. I lost the line breaks and had to add them in by hand. Thank you for all your help.
UC520#sh run | sec voice register
voice register global
mode cme
source-address 192.168.1.100 port 5060
max-dn 160
max-pool 40
load 9971 sip9971.9-4-2SR2-2
authenticate register
authenticate realm all
timezone 13
voicemail 800
tftp-path flash:
file text
create profile sync 7046320445592059
camera
video
voice register dn 1
number 258
name Conf Room
voice register dn 2
number 259
name ACA
voice register dn 3
number 260
name Extension 260
voice register pool 1
id mac 3820.5618.5E20
type 9971
number 1 dn 1
username 258 password *****
description Conf Room
codec g711ulaw
no vad
camera
video
voice register pool 2
id mac 38ED.18E9.6B15
type 9971
number 1 dn 2
username 259 password *****
description ACA
codec g711ulaw
no vad
camera
video
voice register pool 3
id mac AC7E.8A2A.93AC
type 9971
number 1 dn 3
username 260 password *****
description Extension 260
codec g711ulaw
no vad
camera
video
UC520#
Jul 22 20:06:07.234: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:2@192.168.1.100;
user=phone SIP/2.0Via: SIP/2.0/UDP 10.100.20.11:5060;
branch=z9hG4bK2138dfbbFrom: "Extension 260" <sip:260@192.168.1.100>;
tag=ac7e8a2a93ac00150cb714be-6d8b8672To: <sip:2@192.168.1.100>Call-ID: ac7e8a2a-93ac000e-5013cc30-46176608@10.100.20.11Max-Forwards: 70Date: Fri, 22 Jul 2016 20:06:06 GMTCSeq: 101 INVITEUser-Agent: Cisco-CP9971/9.4.2Contact: <sip:260@10.100.20.11:5060;
transport=udp>;
videoExpires: 180Accept: application/sdpAllow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFORemote-Party-ID: "Extension 260" <sip:260@192.168.1.100>;
party=calling;
id-type=subscriber;
privacy=off;
screen=yes
Require: sdp-anat
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-8.0.1Allow-Events: kpml,dialogRecv-Info: conferenceRecv-Info: x-cisco-conferenceAuthorization: Digest username="260",realm="all",uri="sip:2@192.168.1.100;
user=phone",response="631a01b9e50afd4f27a364cb73012692",nonce="98AB7E09005B356A",cnonce="1f8f4221",qop=auth,nc=00000003,algorithm=MD5Content-Length: 629Content-Type: application/sdpContent-Disposition: session;
handling=optionalv=0o=Cisco-SIPUA 17785 0 IN IP4 10.100.20.11s=SIP Callt=0 0m=audio 38590 RTP/AVP 0 102 9 124 8 116 18 101c=IN IP4 10.100.20.11a=rtpmap:0 PCMU/8000a=rtpmap:102 L16/16000a=rtpmap:9 G722/8000a=rtpmap:124 ISAC/16000a=rtpmap:8 PCMA/8000a=rtpmap:116 iLBC/8000a=fmtp:116 mode=20a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=sendrecvm=video 15634 RTP/AVP 97c=IN IP4 10.100.20.11b=TIAS:1000000a=rtpmap:97 H264/90000a=fmtp:97 profile-level-id=42801E;packetization-mode=0;level-asymmetry-allowed=1a=imageattr:* recv [x=640,y=480,q=0.50]a=sendrecv
Jul 22 20:06:07.246: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 420 Bad Extension
Via: SIP/2.0/UDP 10.100.20.11:5060;
branch=z9hG4bK2138dfbbFrom: "Extension 260" <sip:260@192.168.1.100>;
tag=ac7e8a2a93ac00150cb714be-6d8b8672To: <sip:2@192.168.1.100>;
tag=3A8583C-45Date: Fri, 22 Jul 2016 20:06:07 GMTCall-ID: ac7e8a2a-93ac000e-5013cc30-46176608@10.100.20.11CSeq: 101 INVITEAllow-Events: telephone-event
Unsupported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.xContent-Length: 0
07-23-2016 10:36 PM
Can you try this and see if it fixes the problem:
voice service voip
sip
no anat
voice register glo
cre pro
restart
07-25-2016 02:28 PM
Hello Mohammed,
Unfortunately I am still seeing "required: sdp-anat" messages from the phone followed by "420 bad extension" and "unsupported: sdp-anat" from the CME.
I would think there would be a way to modify the phone so that it does require sdp-anat but have not found an option to do this.
Do you think we would have more luck if we concentrate on the phone itself?
Rgds,
Diego
07-25-2016 08:59 PM
Hi Mate,
The response is coming from the phone which indicates that the feature is enabled on the phone.
If the device sends early offer anat will be in supported header but if the device sends delayed offer anat will be in required header.
I know that can be enabled in cucm from sip profile but not able to findout out to enable disable in cme.
Can u try to reset the phone settings in case it is caching this feature from another call control.
On the phone hit settings > administration > reset > reset all
07-27-2016 04:37 AM
Hello Mohammed,
After we did reset all the phone will not register with the CME anymore. I can see it opening the config files using "debug tftp events" but after downloading its config it just sits there showing "not registered" and with a black screen.
In the next few days I am going to bring it into the office to make sure it works in a non-VPN situation and once confirmed it is working 100% in the office I will take it back out to the remote location to see what happens.
I will keep you posted.
Thanks
Diego
07-27-2016 04:53 AM
Can u share debug ccsip messages when the phone is trying to register?
This indicates that the phone wasn't using cme config before the reset and Anat was enabled by old config (probably cucm) as I assumed
07-27-2016 07:58 PM
Below is debug of ccsip messages and also tftp events. You can see that the phone appears to download the cnf file but then it does nothing else. The phone's IP is 10.100.20.10 and it does not show up in any SIP messages. There are some messages from 2.2.2.2 which is the IP I used to replace a public IP for a remote SIP connection we have. I also added the cnf file in hopes that it well help.
Thank you very much for all your help. It is greatly appreciated.
UC520#debug ccsip mess
SIP Call messages tracing is enabled
UC520#
UC520#debug tftp even
Jul 28 02:36:09.693: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3bcff754;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as1833b4fa
To: <sip:192.168.1.100>
Contact: <sip:Unknown@2.2.2.2:5060>
Call-ID: 4aad523965a912e43a19e6f13d01d200@2.2.2.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.4(11.21.2)
Date: Thu, 28 Jul 2016 02:36:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Jul 28 02:36:09.697: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK46565ffd
Max-Forwards: 70
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as54246b31
To: <sip:192.168.1.100>
Contact: <sip:Unknown@2.2.2.2:5060>
Call-ID: 3a04ba664ef56c5943e3977e5571284c@2.2.2.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.4(11.21.2)
Date: Thu, 28 Jul 2016 02:36:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Jul 28 02:36:09.705: //8458/E074C2D08790/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3bcff754;rport
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as1833b4fa
To: <sip:192.168.1.100>;tag=565B5BC-1376
Date: Thu, 28 Jul 2016 02:36:09 GMT
Call-ID: 4aad523965a912e43a19e6f13d01d200@2.2.2.2:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces
Content-Type: application/sdp
Content-Length: 171
v=0
o=CiscoSystemsSIP-GW-UserAgent 196 3892 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.1.100
Jul 28 02:36:09.705: //8459/E0755EF88791/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK46565ffd
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as54246b31
To: <sip:192.168.1.100>;tag=565B5C4-C9F
Date: Thu, 28 Jul 2016 02:36:09 GMT
Call-ID: 3a04ba664ef56c5943e3977e5571284c@2.2.2.2:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resourc
TFTP Event debugging is on
UC520#e-priority,replaces
Content-Type: application/sdp
Content-Length: 171
v=0
o=CiscoSystemsSIP-GW-UserAgent 421 2319 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.1.100
UC520#show debug
TFTP:
TFTP Event debugging is on
CCSIP SPI: SIP Call Message tracing is enabled(filter is OFF)
UC520#
Jul 28 02:37:09.879: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK571b4732;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as3a02b7f2
To: <sip:192.168.1.100>
Contact: <sip:Unknown@2.2.2.2:5060>
Call-ID: 5b83a3f603345c263eef3b7175255bf1@2.2.2.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.4(11.21.2)
Date: Thu, 28 Jul 2016 02:37:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Jul 28 02:37:09.887: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK4a8b9826
Max-Forwards: 70
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as0fdaf700
To: <sip:192.168.1.100>
Contact: <sip:Unknown@2.2.2.2:5060>
Call-ID: 6a792ef718ece5b90dcd2a5f2d38477b@2.2.2.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.4(11.21.2)
Date: Thu, 28 Jul 2016 02:37:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Jul 28 02:37:09.891: //8460/04545BF08792/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK571b4732;rport
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as3a02b7f2
To: <sip:192.168.1.100>;tag=566A0D8-131A
Date: Thu, 28 Jul 2016 02:37:09 GMT
Call-ID: 5b83a3f603345c263eef3b7175255bf1@2.2.2.2:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces
Content-Type: application/sdp
Content-Length: 170
v=0
o=CiscoSystemsSIP-GW-UserAgent 8009 82 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.1.100
Jul 28 02:37:09.895: //8461/045594408793/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK4a8b9826
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as0fdaf700
To: <sip:192.168.1.100>;tag=566A0DC-26E5
Date: Thu, 28 Jul 2016 02:37:09 GMT
Call-ID: 6a792ef718ece5b90dcd2a5f2d38477b@2.2.2.2:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces
Content-Type: application/sdp
Content-Length: 171
v=0
o=CiscoSystemsSIP-GW-UserAgent 547 1560 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.1.100
Jul 28 02:37:36.159: TFTP: Looking for ITLSEPAC7E8A2A93AC.tlv
Jul 28 02:37:47.792: TFTP: Looking for SEPAC7E8A2A93AC.cnf.xml
Jul 28 02:37:47.792: TFTP: Opened flash:/SEPAC7E8A2A93AC.cnf.xml, fd 0, size 3950 for process 157
Jul 28 02:38:10.064: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK496c8cd2;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as745ae10f
To: <sip:192.168.1.100>
Contact: <sip:Unknown@2.2.2.2:5060>
Call-ID: 59f17c3456443b3c0144d73a4c706318@2.2.2.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.4(11.21.2)
Date: Thu, 28 Jul 2016 02:38:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Jul 28 02:38:10.072: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK155c8446
Max-Forwards: 70
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as2c6131ce
To: <sip:192.168.1.100>
Contact: <sip:Unknown@2.2.2.2:5060>
Call-ID: 200cb4084fa650a74afb8f2a5a30d50d@2.2.2.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.4(11.21.2)
Date: Thu, 28 Jul 2016 02:38:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Jul 28 02:38:10.076: //8462/2833F8DF8794/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK496c8cd2;rport
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as745ae10f
To: <sip:192.168.1.100>;tag=5678BF0-51D
Date: Thu, 28 Jul 2016 02:38:10 GMT
Call-ID: 59f17c3456443b3c0144d73a4c706318@2.2.2.2:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces
Content-Type: application/sdp
Content-Length: 171
v=0
o=CiscoSystemsSIP-GW-UserAgent 685 9116 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.1.100
Jul 28 02:38:10.080: //8463/2835312F8795/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK155c8446
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as2c6131ce
To: <sip:192.168.1.100>;tag=5678BF4-2690
Date: Thu, 28 Jul 2016 02:38:10 GMT
Call-ID: 200cb4084fa650a74afb8f2a5a30d50d@2.2.2.2:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces
Content-Type: application/sdp
Content-Length: 172
v=0
o=CiscoSystemsSIP-GW-UserAgent 7607 2299 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.1.100
Jul 28 02:38:12.792: TFTP: Finished flash:/SEPAC7E8A2A93AC.cnf.xml, time 00:00:25 for process 157
Jul 28 02:38:29.445: TFTP: Looking for CTLSEPAC7E8A2A93AC.tlv
Jul 28 02:38:43.521: TFTP: Looking for ITLSEPAC7E8A2A93AC.tlv
Jul 28 02:38:55.194: TFTP: Looking for SEPAC7E8A2A93AC.cnf.xml
Jul 28 02:38:55.194: TFTP: Opened flash:/SEPAC7E8A2A93AC.cnf.xml, fd 0, size 3950 for process 157
Jul 28 02:39:10.246: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK0dcb2d48;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as047ff4d3
To: <sip:192.168.1.100>
Contact: <sip:Unknown@2.2.2.2:5060>
Call-ID: 73b96d1037b0166b20df0c7e59264ad3@2.2.2.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.4(11.21.2)
Date: Thu, 28 Jul 2016 02:39:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Jul 28 02:39:10.250: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK651a44b7
Max-Forwards: 70
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as611b6f58
To: <sip:192.168.1.100>
Contact: <sip:Unknown@2.2.2.2:5060>
Call-ID: 6a1eabd556ec0b917418361318fb5baf@2.2.2.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.4(11.21.2)
Date: Thu, 28 Jul 2016 02:39:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Jul 28 02:39:10.258: //8464/4C12F5D88796/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK0dcb2d48;rport
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as047ff4d3
To: <sip:192.168.1.100>;tag=5687700-1C7E
Date: Thu, 28 Jul 2016 02:39:10 GMT
Call-ID: 73b96d1037b0166b20df0c7e59264ad3@2.2.2.2:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces
Content-Type: application/sdp
Content-Length: 172
v=0
o=CiscoSystemsSIP-GW-UserAgent 7946 9197 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.1.100
Jul 28 02:39:10.258: //8465/4C1392008797/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK651a44b7
From: "Unknown" <sip:Unknown@2.2.2.2>;tag=as611b6f58
To: <sip:192.168.1.100>;tag=5687708-199F
Date: Thu, 28 Jul 2016 02:39:10 GMT
Call-ID: 6a1eabd556ec0b917418361318fb5baf@2.2.2.2:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces
Content-Type: application/sdp
Content-Length: 171
v=0
o=CiscoSystemsSIP-GW-UserAgent 3455 934 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.1.100
Jul 28 02:39:20.194: TFTP: Finished flash:/SEPAC7E8A2A93AC.cnf.xml, time 00:00:25 for process 157
Jul 28 02:39:36.855: TFTP: Looking for CTLSEPAC7E8A2A93AC.tlv
UC520#
UC520#
UC520#di
Jul 28 02:39:48.491: TFTP: Looking for ITLSEPAC7E8A2A93AC.tlv
UC520#more flash:SEPAC7E8A2A93AC.cnf.xml
<device>
<deviceProtocol>SIP</deviceProtocol>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/YA</dateTemplate>
<timeZone>US Eastern Standard Time</timeZone>
<ntps>
<ntp priority="0">
<name>0.0.0.0</name>
<ntpMode>unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<sipPort>5060</sipPort>
</ports>
<processNodeName>192.168.1.100</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<localCfwdEnable>true</localCfwdEnable>
<callForwardURI>service-uri-cfwdall</callForwardURI>
<callPickupURI>service-uri-pickup</callPickupURI>
<callPickupGroupURI>service-uri-gpickup</callPickupGroupURI>
<callHoldRingback>2</callHoldRingback>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>2</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>
<sipStack>
<remotePartyID>true</remotePartyID>
</sipStack>
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>260</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>260</name>
<displayName>Extension 260</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>260</authName>
<authPassword>******</authPassword>
<sharedLine>false</sharedLine>
<messagesNumber>800</messagesNumber>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<enableVad>false</enableVad>
<preferredCodec>g711ulaw</preferredCodec>
<dialTemplate></dialTemplate>
<kpml>1</kpml>
<phoneLabel>Extension 260</phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<dscpForAudio>184</dscpForAudio>
<dscpVideo>136</dscpVideo>
</sipProfile>
<commonProfile>
<phonePassword>260007</phonePassword>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<featurePolicyFile>featurePolicyDefault.xml</featurePolicyFile>
<loadInformation>sip9971.9-4-2SR2-2</loadInformation>
<vendorConfig>
</vendorConfig>
<commonConfig>
<videoCapability>1</videoCapability>
<ciscoCamera>1</ciscoCamera>
</commonConfig>
<sshUserId>260</sshUserId>
<sshPassword>******</sshPassword>
<userId></userId>
<phoneServices>
<provisioning>2</provisioning>
<phoneService type="1" category="0">
<name>Missed Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Received Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Placed Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="2" category="0">
<name>Voicemail</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
<versionStamp>7046320445592054</versionStamp>
<userLocale>
<name>English_United_States</name>
<langCode>en</langCode>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
</networkLocaleInfo>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<servicesURL>http://192.168.1.100:80/CMEserverForPhone/serviceurl</servicesURL>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
</device>
UC520#
Jul 28 02:40:00.663: TFTP: Looking for SEPAC7E8A2A93AC.cnf.xml
Jul 28 02:40:00.663: TFTP: Opened flash:/SEPAC7E8A2A93AC.cnf.xml, fd 1, size 3950 for process 157u all
All possible debugging has been turned off
UC520#
07-28-2016 12:31 AM
Hi,
Do you have the following config? I don't see it in the sh run | sec voice.
voice service voip
no ip add trust authe
all s t s
sip
registrar server
bind all source x/x
Also, try ping as follow from your UC .. ping 192.168.1.100 source 10.100.20.1 (or whatever is your phone gateway)
07-31-2016 11:36 AM
Mohammed,
Yes, all those commands are there. I am going to bring the phone into the office and test it locally. If all goes well I will take it back out to the remote office and keep messing with it. I will keep you posted.
Thanks for all your help,
Diego
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