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Call Failure from SIP client (CME) to Cisco Phone (CUCM)

I have managed to register a third party SIP client on a Cisco Call Manager Express v7.0 successfully. Having done that,  I am able to make calls from the SIP client to the Cisco phones that are registered in the CME and vice-versa as well as calls to the "outside world" through the Call Manager Express Configuration.

However, the following strange thing occurs:

When I am trying to make calls from the SIP client to a Cisco phone that is registered to a Cisco Unified Call Manager (6.1.3 server), the Cisco phone rings, I answer the call on the phone, the phone seems to be connected (counting seconds) but on the SIP client side it keeps waiting for answer .

A couple of seconds later, it seems that the SIP client tries to call again (the Cisco phone accepts a second call) but again with no success until the call is dropped.

The other way around (Cisco of the CUCM to SIP client of CME) works just fine.

Obviously Cisco to Cisco calls between the two Call Managers work normally.

Please see below part of the CME configuration:

voice service voip

no notify redirect ip2ip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

h323

  h225 h245-address on-connect

  no call service stop

  call start slow

sip

  session transport tcp

  registrar server

  midcall-signaling passthru

Do you think that this may be any kind of SIP authentication or configuration that may be missing?

Thanks in advance for any suggestions

11 REPLIES
Cisco Employee

Re: Call Failure from SIP client (CME) to Cisco Phone (CUCM)

It's possible that it may be a codec mismatch. Can you check which dial-peers are matched and what codecs are specified?

-Felipe

Re: Call Failure from SIP client (CME) to Cisco Phone (CUCM)

Hello Felipe,

Thank you for the reply,

From the CME side I have the following (we use a primary and a secondary CUCM)

dial-peer voice 100 voip
description *** CUCM (Primary) ***
destination-pattern 6[1-9]..
session target ipv4:CUCM_IP_address
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs5 media
no vad
!
dial-peer voice 101 voip
description *** CUCM (Secondary) ***
preference 1
destination-pattern 6[1-9]..
session target ipv4:CUCM_IP_address
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs5 media
no vad
!

On the other hand, on the Cisco Unified Call Manager 6.1.3 , while checking on the Region, the codec that is used for the communication between the 2 sites is denoted as G.711.  I am not sure if I need to check something else.

What is really confusing for me is that it works only from the Cisco to the SIP and not the other way around, and most important that when calling from SIP (CME) the Cisco phone (CUCM) rings (that is the call is routed) and shows connected.

Thanks again for your assistance on this

George

Cisco Employee

Re: Call Failure from SIP client (CME) to Cisco Phone (CUCM)

Can you collect the following debug output.

debug voip ccapi inout

debug ccsip message

Please note the calling and called party numbers.

Also, include the full configuration of the gateway.

-Felipe

Re: Call Failure from SIP client (CME) to Cisco Phone (CUCM)

Hello Felipe,

Attached you may find the sh run config of the CME along with the debug file.

Calling Details:

The SIP client with extension 5730 calls the Cisco phone with extension 6638.

CME IP address 10.20.10.10

SIP client:        5730 (IP address 10.1.1.25)

CUCM IP address 10.2.1.250

Cisco phone     6638 (10.2.1.38)

Best regards,

Cisco Employee

Re: Call Failure from SIP client (CME) to Cisco Phone (CUCM)

I'll need you to collect another set of debugs.

debug h225 asn1

debug h245 asn1

debug voip ccapi inout

debug ccsip message

The call is disconnected by CUCM so, if possible, a set of detailed level traces would also help.

-Felipe

Re: Call Failure from SIP client (CME) to Cisco Phone (CUCM)

Hello Felipe,

Please find attached the debug files that you suggested.I hope these work for troubleshooting.

Again I use the same calling scenario:

The SIP client with extension 5730 calls the Cisco phone with extension 6638.

CME IP address 10.20.10.10

SIP client:        5730 (IP address 10.1.1.25)

CUCM IP address 10.2.1.250

Cisco phone     6638 (10.2.1.38)

Please let me know if you think that something is missing,

Thanks once more,

George

Cisco Employee

Re: Call Failure from SIP client (CME) to Cisco Phone (CUCM)

Please upload the CUCM traces. The files collected from CUCM are not the traces but the log files for when the traces were collected. They should either be zipped or in a folder named after the CUCM nodes.

-Felipe

Re: Call Failure from SIP client (CME) to Cisco Phone (CUCM)

I hope these are the correct ones. Please find them attached. In case that these are not helpful I would recreate the test.

Thanks,

George

Cisco Employee

Re: Call Failure from SIP client (CME) to Cisco Phone (CUCM)

make the following changes and test.

On the CME router,

conf t

voice service voip

h323

call start fast

On the h323 gateway page in CUCM,

Uncheck "Wait for far end terminal capability set"

Check "Enable Inbound Fast Start"

Reset the gateway in CUCM

-Felipe

Re: Call Failure from SIP client (CME) to Cisco Phone (CUCM)

I am afraid that this didn't work, as I have the same performance.

voice service voip
no notify redirect ip2ip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
h323
  h225 h245-address on-connect
  no call service stop

sip
  session transport tcp
  registrar server expires max 1200 min 300
  midcall-signaling passthru

I am not sure why to change the settings on the gateway  (even I have done it), since this does not work even internally.

Re: Call Failure from SIP client (CME) to Cisco Phone (CUCM)

Hello Felipe,

Do you have any further suggestions on this? Please let me know if you need additional logs.

kind regards,

George

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