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Transfer between cucm sbc/cube

mmmario99
Level 1
Level 1

Hi everyone! We have a problem with a call transfer.

The structure is simple: My part ((Voip Tlf---->CUCM---->SBC))---->their part---((SBC---->CUCM---->TLF VOIP

When we transfer an internal call (between 2 phones registered in our CUCM) there isnĀ“t any problem, but when we try to transfer a call that comes from sbc/cube we donĀ“t hear the call.

I mean, it works perfectly with the first phone that receives the call, but when this first phone transfer the call to the second phone (from our network too) we cannot hear anything.

What would be the reason? Is there anything I have to enable in the SIP Trunk between SBCs?

We are using G729 for the SBCs' Trunk.

Thanks!!

 

8 Replies 8

Can you post a debug ccsip messages from the components involved in the call transfer for a non-working call? If the call is transferring successfully, but there is no audio my first thought would be a firewall or a NAT problem. But it's hard to say without looking at the debugs.

If you do post the debugs, please put them in one (or more) text files and attach them to your post (rather than pasting the debug into your post). Having the debugs as files makes it easier to look at for troubleshooting.

Maren

I called and try to transfer the call. The phone sound and I when of hook I can not hear anything.

It would help if you could outline what is what in the debug, for example what is the IP(s) of your CMs, what is the IP if the SBC, what is the IP(s) of the ITSP end, what is the calling party number and called party number?



Response Signature


In the log file you have this error "Apr 11 06:38:43.171: %SIP-3-INTERNAL: TCP_SOCKET_SEND_BLOCKED: connid=5755, local_addr=198.192.100.60, remote_addr=198.192.21.8, remote_port=5060 - All retries are exhausted, deleting pending messages"

Looks like you receive the invite from CM for the call on hold for the transfer, but you do not seem to pass that onwards to the ITSP.

Apr 11 06:38:41.086: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:1234561076904@198.192.100.60:5060 SIP/2.0
Via: SIP/2.0/UDP 198.192.73.66:5060;branch=z9hG4bK6b35e2ae612
From: <sip:1233211010000@198.192.73.66>;tag=9672~5db6bb5e-affd-4f12-ba1e-6548401d27db-22807749
To: "PRUEBAS TELEFONICAS" <sip:1234561076904@198.192.100.60>;tag=51181404-1DA5
Date: Thu, 11 Apr 2024 06:38:41 GMT
Call-ID: F37F2484-F70411EE-92B4A378-5EA58731@198.192.100.60
Supported: timer,resource-priority,replaces
Cisco-Guid: 0409166464-0000065536-0000020622-3306455089
User-Agent: Cisco-CUCM15.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Session-ID: 73c18bc200105000a00000aa6e20f81d;remote=87fc1ae28ce8d7a468cc88c7aa585919
Call-Info: <sip:198.192.73.66:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uac
Min-SE:  1800
P-Asserted-Identity: "TRANSFERENCIAS" <sip:1233211010000@198.192.73.66>
Remote-Party-ID: "TRANSFERENCIAS" <sip:1233211010000@198.192.73.66>;party=calling;screen=yes;privacy=off
Contact: <sip:539c03e2-d97e-b7cd-7f03-2efa60918b40@198.192.73.66:5060;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00AA6E20F81D"
Content-Type: application/sdp
Content-Length: 253

v=0
o=CiscoSystemsCCM-SIP 9672 2 IN IP4 198.192.73.66
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:8000
b=AS:24
t=0 0
m=audio 25082 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Would you be able to share your SBC configuration in a file, minus any sensitive information so that we can look at your configuration?



Response Signature


OUR TLF:1233211010000

CUCM -> 198.192.73.66

OUR SBC->198.192.100.60

THEIR SBC ->198.192.21.8

THER TLF:1234561076904

In our case it is a private network and we have no outside lines. The phone to which we want to transfer the call does not appear anywhere

What do you mean by ā€œThe phone to which we want to transfer the call does not appear anywhereā€? I donā€™t understand what you mean by that. Can you please elaborate?



Response Signature


I mean that the debug ccsip does not show the phone to which we want to transfer the call, the dial or information about that phone does not appear. I donĀ“t know if this information should appear.

Thanks!

Itā€™s not expected to be seen as the interaction from your SBC is with their SBC. The SBC acts as the demarcation point between the two networks, so you will not see whatā€™s located on the other side of it.



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