Forwarded call on UC520 is silent

[jon.aril.antonsen]

Hello.

I'm in progress of learning the UC500 series. I have so far got some really good help in here. And I hope I can get som e more.

The UC500 is connected to a SIP trunk with g711alaw support.

The problem now is call's that are forwarded. If i program forwarding of a call with the softbutton on the phone the call is silent. It will set up the call and I can answer the phone, but the call is silent, and there is no dial tone in the calling phone either.

[external phone (1)]------[Public]-----sip-----[UC520]------IP-----[internal phone (2)]

External phone number 51764610

Internal phone number 52223122 / int ext 202

On internal phone i have forwarded to 94849860.

I make a call to 52223122, and the call is forwarded so it starts ringing on the phone with number 94849860.

At this time there is no dial tone on the external phone.

I answer the 94849860 phone.

And then....silence.....

I have tried to hardcode codec on both in an out dial-peer with no luck

Attached is the current running conf and some debugs ccsip mess, ccsip err and voip ccapi inout

I hope someone is able to help me........Dilip are you there ;-)

Regards,

Jon

[Felipe Garrido]

I reviewed the debugs and don't see any problems in the signaling. Can you bring up the forwarded call and collect the following command output?

show call active voice brief

Collect it multiple times while the call is active.

-Felipe

[jon.aril.antonsen]

Hi Felipe.

The requested output is attached.

Jon

[Felipe Garrido]

I've seen the exact problem with another deployment where the SIP SP used AudioCode SBCs. I never got a straight answer from them as they moved the deployment to use an ACME SBC, which resolved the issue.  The only explanation I could come up with is that the SIP SP requires the other endpoint to start transmiting audio before it started transmitting. Since both parties in this scenario are the SP, they never start to transmit. Could you get the SP involved? To verify, can you collect a packet capture off of that outside interface on the UC520?

-Felipe

[jon.aril.antonsen]

Hi.

I can try to talk with the SP. I know that they are using Cisco equipment to, so it might be a chance to work this one out.....

Can the UC520 be configured to start the transmit?

What kind of traffic do you need from the outside? SIP packets?

Jon

[Felipe Garrido]

Just collect an unfiltered packet capture for the call. If they are using Cisco gear, ask them what they are actually using.

-Felipe

[jon.aril.antonsen]

Sorry if this is a stupid question. But how do I capture packets on the UC540?

Jon

[dksingh]

Hmm...u came back for more Jon :-) Guess u like it here...ha ha!

RE: this issue, from what I see there is audio/RTP being Tx/Rx though all signaling looks good...

The provider is using a Cisco/IOS GW as I see following in incoming Invite:

a=rtpmap:100 X-NSE/8000

I think this is more of a SDM/FW or ACL related so wanted to check if u have tried removing

those SDM rules or ACL from FE0/0 and testing?

Though it does not seem to be voice related, just for giggles see if following makes any diff.

voice service voip
sip
   rel1xx disable  <---first try this

   pass-thru content sdp    <---if problem persists, try adding this as well if available in your IOS ver

A packet capture or debug voip rtp pack (very verbose so use with care)  once the call is up will help.

To do a packet capture, try following though this is supported for ISRs only but have seen it working

on other platforms as well:

First configure following:

ip traffic-export profile DKS mode capture

  bidirect
  length 512

interface FastEthernet0/0

  ip traffic-export apply DKS size 2000000

Then from priv/exec mode:

CME-CUE#traffic-export interface fastEthernet 0/0 clear

CME-CUE#traffic-export interface fastEthernet 0/0 start

---setup your call and talk/wait  for 15-20 secs and hang up----

CME-CUE#traffic-export interface fastEthernet 0/0 stop
CME-CUE#traffic-export interface fastEthernet 0/0 copy tftp:

CME-CUE#traffic-export interface fastEthernet 0/0 clear

save the file as .cap on your tftp.

[jon.aril.antonsen]

Hi there Dilip, nice to see you again, and thanks for your reply.

Yep can't get enough of Cisco :-D

I will start testing right away. I will post the result shortly :-)

Jon

[jon.aril.antonsen]

Hi...I'm back :-)

Ok, I tried the different suggestions, and this is the result.

Removing SDM/ACL from interface: No change

Using command rel1xx disable: No change (with or without ACL/SDM on fa0/0)

Addning pass-thru content sdp: No change (with or without ACL/SDM on fa0/0)

The debug voip rtp pack gave no output

But the packet capture worked, the file is attached.

Jon

[dksingh]

Hmm...no output in debug voip rtp pack ?? not good.

But then packet capture does not show any RTP too though sip msgs are there.

Both sides are cisco/IOS so I doubt if there is any need for one side to send RTP before

the other side will start sending anything.

Let's dig a bit deeper if u want....

Pl. capture following in a buffer logging for one call:

deb ccsip all

deb voip ccapi inout

while the call is up:

sh voip rtp connect

sh sccp connect

sh voice call stat

sh sip call

Also, can u confirm that you have codec g711alaw hardcoded on dialpeer 3002 and 1025 ?

And if u remove call-forward all from ephone-dn 11 and answer call on that ip phone, u do

get 2-way audio?

[jon.aril.antonsen]

Hi again.

It took some time waiting for the log to write :-D

The dial-peers have hardcoded codecs.

Show output and debugs are attached :-)

I hope this will help us find a solution :-)

When I remove forwarding I get 2way voice :-)

Jon.

[dksingh]

Thx for the debugs/captures...will look at it in a bit.
Meanwhile, see what happens when you u strip diversion
header off the outgoing invite using following...this is 
just for test and may not yield any diff. results.

voice class sip-profiles 1025
    request INVITE sip-header Diversion remove

 

dial-peer voice 1025 voip

   voice-class sip profile 1025
 

[jon.aril.antonsen]

You were right about that...no change...

I will continue my search for the solution, while I'm waiting for your next reply :-)

Jon

[paolo bevilacqua]

Configure

voice service voip

sip

no supplementaty-service moved-temporarily