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Replies

sdp-anat on CME with 9971 SIP

tato386
Frequent Contributor
Frequent Contributor

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

1 Accepted Solution

Accepted Solutions

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.

View solution in original post

22 Replies 22

Mohammed al Baqari
VIP Advisor VIP Advisor
VIP Advisor

Hi,

ANAT is required for Dual mode support (IPv4/IPv6). Just make sure the preference configured in sip-ua is IPv4 only.

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

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

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

Ok. Can you please share the following.

sh run | sec voice register

debug ccsip messs ---- for a test call.

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

Can you try this and see if it fixes the problem:

voice service voip

sip

no anat

voice register glo

cre pro

restart

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

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

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

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

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#

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)

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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