Re: Can't dial from E20 to from any csico ip phone - Cisco Community

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Dear All,

I would like to seek for help on below issue.

We have Cisco phone 7911, E20 and Tandberg Edge95 and all registered with our CME 7.1.

All is work fine from the past. However, currently one of our E20 is located at other office which is registered with our CME 7.1 via VPN have below problems.

1) E20 can be registered with our CME 7.1 via VPN.

2) E20 can be dial to any number by sip to our office.

However,

1) we can't dial to E20 by SIP sometimes.

What I observed is that Normally, when E20 dial to our main office and then we can dial to E20. After 1 hour, then we can't dial to E20 and it saying " ring out but without ringtone"

We checked the VPN is connected, E20 is registered to our CME 7.1 sucessfully. and also we can abe to ping to E20 device from our CME.

Please let me know if anyone have this issue before and advise how to fix it.

Thank you.

Kenny

11 Replies 11

Try enabling the following debug and then make a call. This should give you some indication as to where the call is failing.

debug ccsip messages

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Dear all,

debug ccsip messages have been enabled as attached. However, I can't see any debug information after made the call as attached. Please advice how to see the debug log. Also when I made the call this morning to E20 device. It said "busy". What is the meaning on this? is it other side enalbe the "DND" button?

You can view the debug output by enabling terminal monitoring. Issue the following command at the privilge exec command prompt.

term mon

After this make the call again and any debugs should be output to the putty session. Make sure your putty scrollback number is high enough to catch all of the output or set putty to log all output to a file.

Dear All,

I see from our firewall log have our CME log as below

51771: Mar 15 01:34:10.288: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2, ConnectionId 1E6C62178C4711E2A118E2004B0D7348, SetupTime 09:33:06.788 PCTime Fri Mar 15 2013, PeerAddress 202, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 09:34:10.288 PCTime Fri Mar 15 2013, DisconnectTime 09:34:10.288 PCTime Fri Mar 15 2013, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0

What is the meaning from above.

The number I called is 202. It only show "ring out" without ringtone.

when I press " acct" button and then ringtone is appeared. what is meaning for "acct" button in our 7911 phone.

Please help and advise.

Thank you

Kenny

Dear All,

After made the call , below is debug output. Is it any thing wrong?

Thank you

Kenny

Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 215

v=0
o=CiscoSystemsSIP-GW-UserAgent 1932 1044 IN IP4 200.200.0.10
s=SIP Call
c=IN IP4 200.200.0.10
t=0 0
m=audio 28182 RTP/AVP 0 19
c=IN IP4 200.200.0.10
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
a=ptime:20

Mar 15 01:53:23.596: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:202@192.168.16.2:5060 SIP/2.0
Via: SIP/2.0/UDP 200.200.0.10:5060;branch=z9hG4bK9F71F71
Remote-Party-ID: "Kenny Hon" <284>;party=calling;screen=no;privacy=off
From: "Kenny Hon" <284>;tag=38606804-17C2
To: <202>
Date: Fri, 15 Mar 2013 01:53:23 GMT
Call-ID: F4223C07-8C4911E2-A16DE200-4B0D7348@200.200.0.10
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 4084563809-2353598946-2708005376-1259172680
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1363312403
Contact: <284>
Call-Info: <200.200.0.10:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 215

v=0
o=CiscoSystemsSIP-GW-UserAgent 1932 1044 IN IP4 200.200.0.10
s=SIP Call
c=IN IP4 200.200.0.10
t=0 0
m=audio 28182 RTP/AVP 0 19
c=IN IP4 200.200.0.10
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
a=ptime:20

Mar 15 01:53:24.596: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:202@192.168.16.2:5060 SIP/2.0
Via: SIP/2.0/UDP 200.200.0.10:5060;branch=z9hG4bK9F71F71
Remote-Party-ID: "Kenny Hon" <284>;party=calling;screen=no;privacy=off
From: "Kenny Hon" <284>;tag=38606804-17C2
To: <202>
Date: Fri, 15 Mar 2013 01:53:24 GMT
Call-ID: F4223C07-8C4911E2-A16DE200-4B0D7348@200.200.0.10
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 4084563809-2353598946-2708005376-1259172680
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1363312404
Contact: <284>
Call-Info: <200.200.0.10:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 215

v=0
o=CiscoSystemsSIP-GW-UserAgent 1932 1044 IN IP4 200.200.0.10
s=SIP Call
c=IN IP4 200.200.0.10
t=0 0
m=audio 28182 RTP/AVP 0 19
c=IN IP4 200.200.0.10
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
a=ptime:20

Mar 15 01:53:26.596: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:202@192.168.16.2:5060 SIP/2.0
Via: SIP/2.0/UDP 200.200.0.10:5060;branch=z9hG4bK9F71F71
Remote-Party-ID: "Kenny Hon" <284>;party=calling;screen=no;privacy=off
From: "Kenny Hon" <284>;tag=38606804-17C2
To: <202>
Date: Fri, 15 Mar 2013 01:53:26 GMT
Call-ID: F4223C07-8C4911E2-A16DE200-4B0D7348@200.200.0.10
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 4084563809-2353598946-2708005376-1259172680
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1363312406
Contact: <284>
Call-Info: <200.200.0.10:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 215

v=0
o=CiscoSystemsSIP-GW-UserAgent 1932 1044 IN IP4 200.200.0.10
s=SIP Call
c=IN IP4 200.200.0.10
t=0 0
m=audio 28182 RTP/AVP 0 19
c=IN IP4 200.200.0.10
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
a=ptime:20

Mar 15 01:53:30.596: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:202@192.168.16.2:5060 SIP/2.0
Via: SIP/2.0/UDP 200.200.0.10:5060;branch=z9hG4bK9F71F71
Remote-Party-ID: "Kenny Hon" <284>;party=calling;screen=no;privacy=off
From: "Kenny Hon" <284>;tag=38606804-17C2
To: <202>
Date: Fri, 15 Mar 2013 01:53:30 GMT
Call-ID: F4223C07-8C4911E2-A16DE200-4B0D7348@200.200.0.10
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 4084563809-2353598946-2708005376-1259172680
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1363312410
Contact: <284>
Call-Info: <200.200.0.10:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 215

v=0
o=CiscoSystemsSIP-GW-UserAgent 1932 1044 IN IP4 200.200.0.10
s=SIP Call
c=IN IP4 200.200.0.10
t=0 0
m=audio 28182 RTP/AVP 0 19
c=IN IP4 200.200.0.10
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
a=ptime:20

Mar 15 01:53:37.148: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId F3757F618C4911E2A168E2004B0D7348, SetupTime 09:53:21.958 PCTime Fri Mar 15 2013, PeerAddress 284, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 09:53:37.148 PCTime Fri Mar 15 2013, DisconnectTime 09:53:37.148 PCTime Fri Mar 15 2013, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
Mar 15 01:53:37.148: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:03/15/2013 09:53:21.960,cgn:284,cdn:,frs:0,fid:27036,fcid:F3757F618C4911E2A168E2004B0D7348,legID:15DE4,bguid:F3757F618C4911E2A168E2004B0D7348
Mar 15 01:53:38.596: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:202@192.168.16.2:5060 SIP/2.0
Via: SIP/2.0/UDP 200.200.0.10:5060;branch=z9hG4bK9F71F71
Remote-Party-ID: "Kenny Hon" <284>;party=calling;screen=no;privacy=off
From: "Kenny Hon" <284>;tag=38606804-17C2
To: <202>
Date: Fri, 15 Mar 2013 01:53:38 GMT
Call-ID: F4223C07-8C4911E2-A16DE200-4B0D7348@200.200.0.10
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 4084563809-2353598946-2708005376-1259172680
User-A\
% Bad IP address or host name
% Unknown command or computer name, or unable to find computer address
sisholdingcme#gent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1363312418
Contact: <284>
Call-Info: <200.200.0.10:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 215

v=0
o=CiscoSystemsSIP-GW-UserAgent 1932 1044 IN IP4 200.200.0.10
s=SIP Call
c=IN IP4 200.200.0.10
t=0 0
m=audio 28182 RTP/AVP 0 19
c=IN IP4 200.200.0.10
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
a=ptime:20

sisholdingcme#term no mon
sisholdingcme#

Dear all,

I just ping cisco E20 device (in other office). The ping time is around 350ms. Would it affect our our Voip IP call to E20 device? Our E20 device firmware is TE2.2.1 and is it it can support by H323 so that i can try to call E20 from our Tandberg edge 95 direct

Thank you.

Kenny

The SIP debug messages you posted show that a call is sent from ext 284 to ext 202. The SIP invite is sent from the CME router to 192.168.16.2 however this device does not respond.

You mention a firewall in a previous post, can you detail what devices are located in the network path between CME and the E20?

Either this packet isn't reaching the E20 (blocked by a firewall) or the E20 can't respond.

Yes, We built the VPN between other office and our office.

cisco phone 7911(200.200.0.36)-->CME 7.1(200.200.0.10)--Firewall(our office200.200.0.1) --VPN by both wan ip--Firewall((other office192.168.16.1)--> E20 device ( 192.168.16.2)

So other office is always can direct to us. I cannot dial to them except when they dial to me then i can dial to them. After a while or 1 hour later, I cannot dial to them.

Then i check with our CME server and the device is show registered by show voice register pool 2 ( the no registered in our CME). Is it the firewall block our incoming packet and only allow outgoing packet go to our server? what is the port they block?

How to test if their firewall block the incoming packet?

Thank you

Kenny

What is the model of your firewall?

If your firewall is trying to perform SIP ALG functions on the packet, this might also explain why the call setup fails. If SIP ALG is enabled try turning it off.

SIP uses port 5060 and can be either TCP or UDP (by default it is UDP). Your debugs show that your system is using UDP.

Dear

my office is using Fortinet 110C and other office I guess using netscreen model.

I just try to telnet to E20 device by port 5060. It can go through to E20 device.

Is it both side of firewall need to disable SIP ALG function?

Thanks

Kenny

I would disable it on both ends to rule it out.

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