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SIP Call Transfer from CCM

antonio.fragoso
Level 1
Level 1

Hi Team

I have  a SIP  trunk between my CCM ver 6 with my Voice Gateway .

I can make a call normaly , but I am not able to transfer to another CCM   IP Phone

Please  any sugestion ?

1 Accepted Solution

Accepted Solutions

You said you have CUCM not CME, so this is not relevant here.

You also said you are using SIP for the call control, looking at your config you are using H323 between this GW and CallManager:

dial-peer voice 300 voip

destination-pattern .

voice-class codec 2

session target ipv4:172.19.0.253

you may want to add the following to the dial-peer:

dial-peer voice 300 voip

voice-class codec 1  2 as you have 2 of them defined, not sure what you wanted to use

no vad

The debug appears to be incomplete, did you post it for the entire call?

Chris

View solution in original post

18 Replies 18

Chris Deren
Hall of Fame
Hall of Fame

So, what are you getting when you attempt to transfer the call?  Can you post "debug ccsip messages" for the failed call?

Is the other IP phone at the same location using the same device pool?

Chris

HI Chris

When I transfer the Call on the destination  Phone only apear my Phone Number I  not able To hear anything

I have Otcher CCME connect to the same Voice Gateawy  using call ( but not SIP ) and the transfer work fine.

Yes the IP Phone are on the same location and using the same device pool

So, call control appears fine, but no voice?

Are you binding SIP protocol to proper interface?

Chris

What you means bay Call COntrol ?and the SIP is correct configured

The Park OPtions is working fine

Regards

You say the issue is with not being able to hear the voice, yet the phone displays the calling name, this tells us the call control between all devices is working, but the voice stream does not get to where it needs. Typically this is either network routing issue, or protocol binding.

Make sure that on your voice gateway you are binding SIP protocol, here is an example:

voice service voip

sip

  bind control source-interface Port-channel22

  bind media source-interface Port-channel22

Chris

THe situation is:

   - I call an ontside number througt SIP trunk - The call is OK

   - I push the "transfer" key and dial the internal number to  transfer

   - The other site pick up my call , and I dial "trnasfer" to complet the call transfer

   - On the other site keep show my oun number

I don have the voice service  voip as you  say . I just have the sip-ua configured

Please post your full gateway (CUBE) configuration.

Chris

Hi Chris

Here you have

Regards

Notice you are not binding SIP to any interface, do this:

voice service voip

sip

  bind control source-interface GigabitEthernet0/1

  bind media source-interface GigabitEthernet0/1

Also, I slightly confused with what you have on this gateway as you appear to be running H.323, MGCP and SIP, which connection is for what?

Chris

Hi Chris

Still the same

If I put the "bind control"  the "bind media" desapear

The H.323 and MGCP are used for other devices  under the other CCME tha use this Gateway

Regards

OK, is this the call flow:

SIP trunk -- GW (CUBE) -- CUCM -- IP Phone A --> transfer to IP Phone B?

What is the IP address of the CUCM, as I do not see any SIP dial peers pointing to CUCM, only H323 dial-peers.

Can you post "debug ccsip messages" for the failed transfer?

Chris

Hi Antonio

Take a look to this

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrans.html

Look for the call transfer methods

.

These enhancements are supported only if:

The transfer-system full-consult command (default) is set in telephony-service configuration mode.

The transfer-mode consult command (default) is set for the transferor's directory number (ephone-dn).

An idle line or channel is available for seizing, digit collection, and dialing.

Here you have th edebug

The CUCM IP  172.19.0.253

HI  Rafael

I this  what you suggest it for CME

And the link you sugest its not  available

Regards

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