CUCM troubleshooting, can i tell if CUCM is receiving a call?

[twhittle1]

Hello,

I'm trying to better understand and troubleshoot my CUCM environment. I have CUCM 11.5, a CUBE router and an ITSP SIP provider as my connection to the PSTN. 

I have outbound calls working, internal numbers can call and voice calls work fine, however I cannot get incoming calls to work. 

I believe my CUBE router is forwarding calls to CUCM because when I do debug voice ccapi inout the output shows the correct inbound and outbound dialpeers being matched.

So I believe my cube is telling me it is forwarding the calls correctly but when I call from my mobile (cell) phone inwards the phone rings and eventually goes to the the SIP providers voicemail.

Are there any commands I can run on the CUCM to troubleshoot this? To see if CUCM is receiving the call? And then if it is receiving the call, why its not connecting it?

Many Thanks,

Tom

[twhittle1]

Hi,

I've not added any CAC, I'll have a read through and see if I can add some CAC.

Thanks,

Tom

[twhittle1]

Hi,

I've just read through and added  max-connection 5 to each dial-peer and additionally call threshold global total-calls low 1 high 10 and so far no change :(

Thanks,

Tom

[Moshfiqur Rahman]

Hi Tom,

you can modify  "dial-peer voice 1000 voip" to receive incoming call from CUCM

dial-peer voice 1000 voip
 description CUBE-TO-CUCM
 destination-pattern 2..
 incoming called-number 2..
 session protocol sipv2
 session target ipv4:172.16.127.70
 codec g711ulaw
 no vad

but this one shouldn't be cause of the error code "480 Temporarily Unavailable (Call limit)".
This error message coming from CUCM to CUBE, Looks like it is Call Admission Control related issue. I will test in my home lab after office hour and will see if I can replicate the error or not.

Regards,

Moshfiqur

[Moshfiqur Rahman]

Hi Tom,

I am so sorry, I didn't see that the CUBE has sent second invite to CUCM.
so, digest authentication doesn't matter in this case. In response to second invite from CUBE , CUCM is responding back with

"SIP/2.0 480 Temporarily Unavailable (Call limit)"

Is there any location applied in the trunk itself or in the device pool that is being applied in this trunk? If so, Can you make location setting as "Hub-None" and test?

[twhittle1]

Hi,

Thanks for continuing to look at this. The location on the trunk is set to Hub_None, see attached

what else might cause the call limit?

Many Thanks,

Tom

[HARIS_HUSSAIN]

Dear twhittle,

As per your logs 

CUBE Tries to Establish connection with CUCM but 

CUCM Says

SIP/2.0 480 Temporarily Unavailable (Call limit)

Are you having any Bandwidth or Location settings.

Make sure the Location setting on SIP Trunk allow enough Calls to pass through to the Device.!!

Instead of 201 extension can you try some other extension as well.

Thanks

Haris

>>Please rate and mark correct as applicable

[twhittle1]

Hi,

Interestingly I'm not getting the Call Limit error any more. Attached is the revised debug ccsip messages. 

Can you anything else which stands out as strange in the debug?

Thanks,

Tom

[twhittle1]

Hi All,

I've gotten somewhere! I removed the "outbound proxy" from my voice service voip command and now inbound calling is working.

I can't fully explain it but it seems like this command was catching all the sip calls and redirecting them to the proxy?

I'm having one more issue where my incoming call is not disconnecting when the calling mobile phone hangs up (internal phone keeps ringing) but this is something I'll investigate seperately.

Progress!

Thanks for your assistance.

Tom

[HARIS_HUSSAIN · Accepted solution]

Great it worked out , Outbound proxy command is used to route  SIP Messages to the outbound proxy . You can even use outbound proxy on individual dial-peers.

Please rate if found applicable.

Thanks

Haris

[twhittle1]

Ah ok, so if I configure it in the voice service voip area it applies globally or it can be applied specifically to a dial peer. That makes sense, I had it in the wrong place and it was catching and redirecting all sip calls to my ITSP rather than just my outbound calls.

As it turns out I don't appear to need it but it's very good to know for the future!

Thanks again for your help.

Regards,

Tom

[Moshfiqur Rahman]

Good catch Tom! happy to know that your issue has been resolved.

[twhittle1]

Hi,

I'm trying to find other useful information. Attached is the debug voice ccapi inout. I'm not very familiar with debugging voice yet but I can't see anything too off.

Thanks,

Tom

[twhittle1]

Hi,

A couple of other things I've noticed:

The when the debug ccsip messages command is run and I place a call it seems to hang while the mobile phone rings:

Ro-H-CUBE#SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.127.15:5060;branch=z9hG4bKB990F;received=172.16.127.15;rport=54414
From: "07000111222" <sip:07000111222@phone.tel2.co.uk>;tag=93A9EC-1D94
To: <sip:201@172.16.127.70:5060>
Call-ID: 504BD4C0-68B211E6-82E5A61E-12822F40@172.16.127.15
CSeq: 102 INVITE
Server: 2talk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:201@27.111.15.65:5060>
Content-Length: 0

It seems to be the CUCM isn't responding?

Also I notice all the entries say SIP/2.0/UDP - i assume using UDP rather than TCP. Is there a default setting anywhere in CUCM so that CUCM is expecting TCP? Could this be causing an issue?

I wish I understood this better, it's how you learn I guess, just frustrating whilst I can't get thing working :)

Thanks again,

Tom