04-30-2014 07:59 AM - edited 03-16-2019 10:38 PM
Hi everyone,
I am currently working on a Cisco 2911 running 15.2(4)M5, which is version 9.1 of CME. I am having a problem making outbound calls from the 9971 phone that I have registered. When I dial, the phone just sits there and eventually times out with a fast busy after a few minutes. I currently have the "video" and "camera" commands configured under "voice register global" and "voice register pool" to enable video on the phone.
I have found that when those two commands are removed, I am able to make outbound calls from the phone. I am wondering if this is a bug in the version of CME I am running or if it is simply a configuration error on my part somewhere.
For your reference, here is a configuration snippet:
voice register global
mode cme
source-address 10.8.88.253 port 5060
max-dn 20
max-pool 10
authenticate register
timezone 8
url authentication http://10.8.88.253/CCMCIP/authenticate.asp
tftp-path flash:
create profile sync 0014032233425515
ntp-server 67.217.112.181 mode directedbroadcast
camera
video
voice register dn 5
number 7005
name Office_5
label Line 1 - 7005
voice register pool 5
id mac 1C1D.86C5.42E3
type 9971
number 1 dn 5
dtmf-relay sip-notify
username cisco password cisco
codec g711ulaw
camera
video
Solved! Go to Solution.
05-01-2014 07:44 AM
It definitely sounds like a bug to me. I no longer have access to internal tools but did an external search for you.
It looks like you're hitting CSCty61190
That is just a duplicate of CSCtt38880 (https://tools.cisco.com/bugsearch/bug/CSCty61190)- IOS gateway not handling fragmented SIP UDP message properly
So it looks like the problem is due to the extra video SDP information causing the packet to be big enough to get fragmented and IOS not handling the fragmented message appropriately.
04-30-2014 08:03 AM
If outbound as in PSTN, you failed to mention what you use to connect to the PSTN.
I'd assume a PRI, if so, this has been covered many times before in CSC.
You need bearer-cap speech command.
04-30-2014 08:13 AM
I am referring to internally dialing only--PSTN is not involved at this point. For example, I am trying to call a SCCP phone configured under telephony-service with the following configuration:
telephony-service
no auto-reg-ephone
max-ephones 10
max-dn 20
ip source-address 10.8.88.253 port 2000
service phone webAccess 0
time-zone 8
max-conferences 8 gain -6
web admin system name administrator secret 5 $1$zwBQ$4UMNn8j1BYoxsLC8FRP3f1
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Apr 30 2014 22:31:49
ephone-dn 9
number 7019 no-reg primary
label Admin_9th
name Line 1 - 7019
ephone 1
mac-address 1C1D.862F.2F24
type 7965
button 1:9
04-30-2014 08:35 AM
Can you grab a "debug ccsip messages" and "debug ephone detail" for one of the calls?
04-30-2014 08:47 AM
Hi Brian,
There were no logs to gather:
1. Call placed from 9971 @ 00:00:00
2. Phone times out at 01:01:00
3. No messages received in "debug ccsip messages" or "debug ephone detail"
04-30-2014 09:21 AM
Do you have "term mon" enabled?
What does your "show logging" output show? Is monitor logging enabled at debug level?
04-30-2014 09:24 AM
Yes, Brian. I am able to see debug output since I had term mon enabled. I'm just not getting anything from the phone.
06-15-2014 03:03 PM
Andy,
Its very interesting well have you configure any command passthr content sdp on sc router under voice service voip if yes then you will even not able to make outbound calls from SIP phones. But yes from SCCP
I am reading your post --- and you have stated ( Once I set the MTU to a number at or above 1504, I can now make calls. ) where did you changed the MTU can you please explore!
Regards
Arshad
04-30-2014 09:46 AM
I enabled SSH on the phone and enabled the following debugs:
debugs: sip-adapter cc-msg sip-task sip-state sip-messages sip-reg-state sip-trx timers ccdefault call-event
Attached is the log from a test call that I just made. I am going through it now, but it would be very helpful to get a second set of eyes on it if you have the time. Thanks in advance.
04-30-2014 09:51 AM
It looks like the phone is sending the Invite but never getting a response back from CME so it keeps resending the Invite over and over:
2073 DEB 00:37:23.007137 CVM-sipio-sent---> INVITE sip:7019@10.8.88.253;user=phone SIP/2.0^M
Via: SIP/2.0/UDP 10.8.97.34:5060;branch=z9hG4bK5391136b^M
From: "Office_5" <sip:7005@10.8.88.253>;tag=1c1d86c542e30008699f2da7-39437136^M
To: <sip:7019@10.8.88.253>^M
Call-ID: 1c1d86c5-42e30005-43edaaa0-5f38f18d@10.8.97.34^M
Max-Forwards: 70^M
Date: Wed, 30 Apr 2014 16:37:22 GMT^M
CSeq: 101 INVITE^M
User-Agent: Cisco-CP9971/9.3.4^M
Contact: <sip:982D0E4-1D76@10.8.97.34:5060;transport=udp>;video^M
Expires: 180^M
Accept: application/sdp^M
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO^M
Remote-Party-ID: "Office_5" <sip:7005@10.8.88.253>;party=calling;id-type=subscriber;privacy=off;screen=yes^M
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-6.0.2,X-cisco-xsi-8.0.1^M
Allow-Events: kpml,dialog^M
Content-Length: 622^M
Content-Type: application/sdp^M
Content-Disposi
2074 DEB 00:37:23.007267 CVM-tion: session;handling=optional^M
^M
v=0^M
o=Cisco-SIPUA 2312 0 IN IP4 10.8.97.34^M
s=SIP Call^M
t=0 0^M
m=audio 29052 RTP/AVP 0 8 18 102 9 116 124 101^M
c=IN IP4 10.8.97.34^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:18 G729/8000^M
a=fmtp:18 annexb=no^M
a=rtpmap:102 L16/16000^M
a=rtpmap:9 G722/8000^M
a=rtpmap:116 iLBC/8000^M
a=fmtp:116 mode=20^M
a=rtpmap:124 ISAC/16000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-15^M
a=sendrecv^M
m=video 20310 RTP/AVP 97^M
c=IN IP4 10.8.97.34^M
b=TIAS:1000000^M
a=rtpmap:97 H264/90000^M
a=fmtp:97 profile-level-id=42801E;packetization-mode=0;level-asymmetry-allowed=1^M
a=imageattr:* recv [x=640,y=480,q=0.50]^M
a=sendrecv^M
04-30-2014 10:04 AM
That's what I'm seeing as well Brian. I also found this line interesting:
2170 DEB 00:38:26.502042 CVM-SIPCC-SIP_PROXY: ccsip_pick_a_proxy: Unable to reach proxy, attempting backup.
The question is: Why is CME not responding? And why am I not receiving any output from the "debug ccsip messages" command? I just did a "debug ip packet detail" for an access list only including SIP and got 7 packets...I'm thinking that those were the invites. However, they just don't show up in the SIP debug. I even enabled the "debug ccsip all" and still got nothing.
04-30-2014 10:11 AM
One thing I noticed is the phone is sending the SIP traffic over UDP. It's possible CME isn't listening on UDP port 5060.
What does "show control-plane host open-ports" show?
04-30-2014 10:23 AM
Brian,
You are correct, CME is currently NOT listening on UDP 5060:
CME-RTR#show control-plane host open-ports | i 5060
tcp *:5060 *:0 SIP LISTEN
tcp *:5060 *:0 SIP LISTEN
04-30-2014 10:26 AM
Very strange. I'm not sure if there's any config that would control that. You could try removing the source-address under voice register global and re-adding it then running the command again to see if it then listens on UDP 5060 as well.
04-30-2014 10:36 AM
Hi Brian,
I went ahead and disabled video to see if the control plane information changed. Everything still looks the same--not listening on UDP 5060, but the call goes through successfully.
Here's the initial invite:
Received:
INVITE sip:7019@10.8.88.253;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.8.97.34:5060;branch=z9hG4bK05a6911c
From: "Office_5" <sip:7005@10.8.88.253>;tag=1c1d86c542e3000716298913-2e68363d
To: <sip:7019@10.8.88.253>
Call-ID: 1c1d86c5-42e30005-6693906b-076639fa@10.8.97.34
Max-Forwards: 70
Date: Wed, 30 Apr 2014 17:33:23 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP9971/9.3.4
Contact: <sip:BEC0-1B6A@10.8.97.34:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Office_5" <sip:7005@10.8.88.253>;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-6.0.2,X-cisco-xsi-8.0.1
Allow-Events: kpml,dialog
Content-Length: 400
Content-Type: application/sdp
Content-Disposition: session;handling=optional
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