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CME 4.1 to CUCM 5.1 SIP Trunk

wbarren
Level 1
Level 1

I have setup a sip trunk between my call manager and cme. I can call any phone from CUCM to the CME. When I try to place a call from CME to CUCM I am getting a fast busy. Here is by SIP Trunk on the CME side

dial-peer voice 1003 voip

destination-pattern 393

session protocol sipv2

session target ipv4:x.x.x.x

session transport tcp

dtmf-relay rtp-nte

codec g711ulaw

no vad

transfer-system full-consult

transfer-pattern .T

transfer-pattern 8...

transfer-pattern 128..

transfer-pattern 168..

I currently have the destination setup to try and call 1 specific phone for troubleshooting. On my CUCM side I have CSS set, Destination Address, MTP Preferred Origination Codec 711ulaw. Any ideas?

1 Accepted Solution

Accepted Solutions

Just noticed you have "session transport tcp". Please remove that to leave the default of UDP.

View solution in original post

8 Replies 8

Ayodeji Okanlawon
VIP Alumni
VIP Alumni

Please check the following:

1. MRGL that yout sip trunk is assigned has a software MTP in it

2. The CSS on the inbound calls on your sip trunk has the partiton of your ip phones in it

Please rate all useful posts

I have confirmed that the SIP trunk:

MRGL has a software MTP in it

CSS for inbound calls is in the partition of my phones.

Hi, "393" doesn't seem right as destination-pattern. Suppose CM extensions all start with 3 and are four digits, "3..." would be right.

In any case, take "debug ccsip message" with "term mon" when calling. That will show numbers and causes.

All the ext on the CUCM side are 3 digits. I have the destination-pattern set for 393 because I was testing with 1 extention for troubleshooting.

Then collect the debug trace as mentioned above.

Just noticed you have "session transport tcp". Please remove that to leave the default of UDP.

That worked thanks!

Thanks for the appreciation and good luck!

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