In Cisco CallManager 7.x

[kalbasha0]

When an net call is attempted through the Session Initiation Protocol (SIP) trunk, the caller gets a fast busy tone. All the incoming calls to the IP phones are fine.

Any idea

pls help

[srsivara]

Kyle,

Who's the SIP provider ?

- Does the outbound call from any IP phone within the CallManager network fail ? If so, you can try toggling the Outbound Transport Type on the SIP Security Profile (System > Security > SIP Security Profile) associated with the SIP trunk, reset the SIP trunk, and make another test call

- If there a CUBE involved in the call flow, you can run 'debug ccsip mess' on the CUBE, make an outbound call and check if you the SIP INVITE coming from the CallManager

- If the SIP trunk goes from the CallManager directly to the SIP provider, you can contact the SIP provider and check if they are receiving the SIP INVITE.

- Might also be worth checking the SIP provider requires early media (SIP INVITE with SDP). That would require checking the 'Media Termination Point Required' on the SIP trunk and ensuring that the SIP Trunk's MRGL has access to an available MTP

- You can also run the command 'utils network capture port ' on the SSH session to the CallManager servers in the SIP trunk's DP, and check if there is an outbound connection / msg being sent when an outbound call is made. By default, would be 5060 for non-secure SIP trunks.

- Sriram

[kalbasha0]

Prior to real deployment, I'm in the process of testing, and what I have done is the following:

Setup a lab environment and configure SIP trunk utilizing non secure SIP Trunk profile

Between our LAB server and our publisher

Publisher and LAB

I was able to conduct a-net call between phones a registered with our publisher to a phone B registered with our lab server,

However whenever I tried to make a phone call from LAB phone B into phone A registered with our publisher, I received a fast busy

thx

[srsivara]

Can you check if the 'Incoming CSS' on the Publisher's SIP trunk has access to Phone A's partition ?

[kalbasha0]

The SIP CSS on the publoisher has access to phone A pertition

[srsivara]

Is the 'Significant Digits' on the Publisher's SIP trunk set correctly to match Phone A's directory

number ?

[kalbasha0]

Yes it's

If you got time you may call me @ 9497548528

[srsivara]

Can you do the following ?

- Set the CallManager traces to detailed :

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094e89.shtml#calm

- Make a test call and collect the 'Cisco Callmanager' traces using RTMT

- Use text editor like notepad++ to search through the sub-folders for the following string

INVITE sip:

- If you get any result with the To header having your phone A's number, paste it here

- Sriram

[kalbasha0]

The Tace should be on the pub or lab server ?

pls advise

thx

[srsivara]

Actually, you can set it on both servers, download the 'Cisco CallManager' servers. Make sure that the 'SIP Call Processing' and 'SIP Stack Trace' options are also checked for the trace settings.

You can search for 'INVITE sip:' on the Publisher, and see if you see any incoming SIP messages from the Lab server for Phone A's number

- Sriram

[kalbasha0]

I'm not sure how to view it via RTM. Since I'm using window

I choose the following

System \ tools \ trace log\ collect file \ ccm \ ?

I'm not sure what  system services application I choose

pls advise

thx