01-23-2012 07:52 AM - edited 03-16-2019 09:09 AM
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 ?
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01-24-2012 10:33 AM
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
no vad
The debug appears to be incomplete, did you post it for the entire call?
Chris
01-23-2012 08:09 AM
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
01-23-2012 08:16 AM
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
01-23-2012 01:46 PM
So, call control appears fine, but no voice?
Are you binding SIP protocol to proper interface?
Chris
01-24-2012 05:31 AM
What you means bay Call COntrol ?and the SIP is correct configured
The Park OPtions is working fine
Regards
01-24-2012 05:52 AM
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
01-24-2012 07:58 AM
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
01-24-2012 08:00 AM
Please post your full gateway (CUBE) configuration.
Chris
01-24-2012 08:22 AM
01-24-2012 08:48 AM
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
01-24-2012 09:17 AM
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
01-24-2012 09:26 AM
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
01-24-2012 09:39 AM
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.
01-24-2012 09:45 AM
01-24-2012 09:58 AM
HI Rafael
I this what you suggest it for CME
And the link you sugest its not available
Regards
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