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New Member

CME 9971 with Video Enabled Cannot Make Calls

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

Everyone's tags (4)
1 ACCEPTED SOLUTION

Accepted Solutions

It definitely sounds like a

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

Symptom: When video/camera is used on the 9971 phones, dialed digits are not recognized and calls do not work.

 

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.

20 REPLIES
Cisco Employee

If outbound as in PSTN, you

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.

HTH

java

if this helps, please rate

www.cisco.com/go/pdi
New Member

I am referring to internally

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

 

 

Can you grab a "debug ccsip

Can you grab a "debug ccsip messages" and "debug ephone detail" for one of the calls?

New Member

Hi Brian,There were no logs

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"

Do you have "term mon"

Do you have "term mon" enabled?

 

What does your "show logging" output show?  Is monitor logging enabled at debug level?

New Member

Yes, Brian.  I am able to see

Yes, Brian.  I am able to see debug output since I had term mon enabled.  I'm just not getting anything from the phone.

New Member

Andy,Its very interesting

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

 

New Member

I enabled SSH on the phone

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.

It looks like the phone is

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

New Member

That's what I'm seeing as

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.

One thing I noticed is the

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?

New Member

Brian,You are correct, CME is

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

Very strange.  I'm not sure

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.

New Member

Hi Brian,I went ahead and

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

New Member

Hi Brian,Any other thoughts

Hi Brian,

Any other thoughts on this?  I'd really like to figure out what's going on.  Do you think it's a bug?  Mind doing an internal search for me?

Thanks for your help!

It definitely sounds like a

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

Symptom: When video/camera is used on the 9971 phones, dialed digits are not recognized and calls do not work.

 

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.

New Member

The keyword being FRAGMENTED

The keyword being FRAGMENTED message.  Right now I am actually running these phones through a dot1q tunnel and neglected to set the MTU correctly on one of the switches that is processing the frames.  Once I set the MTU to a number at or above 1504, I can now make calls. 

Thank you for your help with this Brian!  I appreciate your efforts.

I think CME may not be

I think CME may not be accepting the call due to no video bandwidth configured.

 

Try this:

voice register global

 mode cme

 bandwidth video tias-modifier 512000 negotiate end-to-end

New Member

Hi Brian,That sounded

Hi Brian,

That sounded promising, but it didn't seem to like that either.  I went into "voice register global", entered the "bandwidth video" command as stated above, did a "create profile", reset the voice register pool, and still got the same result.

New Member

By the way, I can make a

By the way, I can make a successful call if it originates from the SCCP phone.  So we know that the SIP phone can successfully receive calls, it just cannot place them at this point.

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