SIP Trunk with CUBE

[moath.almogbel]

Hi,

We have  CUCM 8.6.2 and 3945 router with MGCP controlled PRI's for PSTN Access. We need to connect also to a new provider using SIP

for incoming/outgoing SIP calls.

So I have configured CUBE as per the documents available but cannot place an outgoing call using SIP to the SIP Provider.

The setup is:  CUCM---> SIP Trunk ---> CUBE ---> SIP ---> SIP Provider.

I am new to CUBE so I am attaching ccsip debug and ccapi debug, also the sh run.

any help is appreciated.

Thanks!

[Chris Deren · Accepted solution]

You are not getting anything back from your SIP provider, can you ping 80.90.160.221? Are you sure this is the Proxy IP address? Does the trunk perhaps require authentication?

Chris

[moath.almogbel]

Thank you Chris for your reply.

Yes I am able to ping it and this is the address I got from the provider and they told me they authenticate only using the IP address.

VG-1#ping 80.90.160.221

Type escape sequence to abort.

Sending 5, 100-byte ICMP Echos to 80.90.160.221, timeout is 2 seconds:

!!!!!

Success rate is 100 percent (5/5), round-trip min/avg/max = 1/2/4 ms

By the way, I am connected to the provider throught a WAN (not point to point) does this make any difference?

Also can you show me the part of the trace the tells I am not getting any thing from provider so that I can raise it with them.

One more thing, did you find my config is correct? is there any advices/changes that I shoud consider.

Thanks Alot for your help

[Ayodeji Okanlawon]

Moath,

Here is what I see...

You have a sip trunk configured from CUCM to your gateway. But your inbound dial-peer for the called number is h323...

dial-peer voice 10 voip

incoming called-number 0796260696

This wont work and thats why you are getting network out of order..

To resolve this configure this..

dial-peer voice 10 voip

session protocol sipv2

incoming called-number 0796260696

dtmf-relay rtp-nte

no vad

Do another test call and if yous till have issues please send only debug ccsip message and debug voip ccapi inout. Do not send debug ccsip all..If we need it we will ask later..This is because it is truncating your debugs.

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[moath.almogbel]

Hi aok,

Thank you for your reply,

I have chnaged the dail-peer 10 as you suggested, but still cannot place the call.

attached are the requested debugs.

I really appreciate your help.

[Ayodeji Okanlawon]

Ok,

Lets try this..remove the mode-border element command..you already have allow-connections sip to sip

voice service voip

  no mode border-element

Then send me debug ccsip messages again

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[moath.almogbel]

Here you go,

Feb  1 23:15:57.587: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:0796260696@10.0.128.11:5060 SIP/2.0

Via: SIP/2.0/TCP 10.1.3.12:5060;branch=z9hG4bK114750d9e1fc

From: "PHONE B" <8351>;tag=123088~bda7d047-7f13-4d7c-84b8-3fccaa8c92b0-61321008

To: <0796260696>

Date: Fri, 01 Feb 2013 23:20:41 GMT

Call-ID: fd5bf000-10c14dc9-630-c03010a@10.1.3.12

Supported: timer,resource-priority,replaces

Min-SE:  1800

User-Agent: Cisco-CUCM8.6

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

CSeq: 101 INVITE

Expires: 180

Allow-Events: presence, kpml

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Call-Info: <10.1.3.12:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

Cisco-Guid: 4250660864-0000065536-0000000926-0201523466

Session-Expires:  1800

P-Asserted-Identity: "PHONE B" <8351>

Remote-Party-ID: "PHONE B" <8351>;party=calling;screen=yes;privacy=off

Contact: <8351>;video;audio;video

Max-Forwards: 69

Content-Length: 0

Feb  1 23:15:57.591: //1/FD5BF0000000/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 10.1.3.12:5060;branch=z9hG4bK114750d9e1fc

From: "PHONE B" <8351>;tag=123088~bda7d047-7f13-4d7c-84b8-3fccaa8c92b0-61321008

To: <0796260696>

Date: Fri, 01 Feb 2013 23:15:57 GMT

Call-ID: fd5bf000-10c14dc9-630-c03010a@10.1.3.12

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Feb  1 23:15:57.603: //1/FD5BF0000000/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/TCP 10.1.3.12:5060;branch=z9hG4bK114750d9e1fc

From: "PHONE B" <8351>;tag=123088~bda7d047-7f13-4d7c-84b8-3fccaa8c92b0-61321008

To: <0796260696>;tag=28AD8-A27

Date: Fri, 01 Feb 2013 23:15:57 GMT

Call-ID: fd5bf000-10c14dc9-630-c03010a@10.1.3.12

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=38

Content-Length: 0

Feb  1 23:15:57.603: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:0796260696@10.0.128.11:5060 SIP/2.0

Via: SIP/2.0/TCP 10.1.3.12:5060;branch=z9hG4bK114750d9e1fc

From: "PHONE B" <8351>;tag=123088~bda7d047-7f13-4d7c-84b8-3fccaa8c92b0-61321008

To: <0796260696>;tag=28AD8-A27

Date: Fri, 01 Feb 2013 23:20:41 GMT

Call-ID: fd5bf000-10c14dc9-630-c03010a@10.1.3.12

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: presence, kpml

Content-Length: 0

[Ayodeji Okanlawon]

Ok,

CUBE is sending service unavailable...

Lets tidy up a few things... ( i want us to use the same transport protocol and change our dtmf-relay config)

dial-peer voice 11 voip

no session transport tcp---remove this

no dtmf-relay rtp-nte h245-alphanumeric-----------remove this

dtmf-relay rtp-nte---------add this

send debugs again..if it doesnt work then test again and send debug ccsip all

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[moath.almogbel]

Hi aok,

I have updated the dial-peers but still same issue.

attached is the ccsip all debug.

One question, could it be a codec mismatch thing?

[Ayodeji Okanlawon]

Moath,

It is not a codec issue. You have a Socket problem on UDP protocol and I assume it is the same with tcp. The gateway cannot establish a soccket connection to your ITSP. This is why it cant even originate the Invite to it.

What we will do now is to swicth back to tcp and then send me the debug ccsip all again.

dial-peer voice 11 voip

session transport tcp

Feb  2 00:12:52.323: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWSocketException: context=0x13C57EC0

Feb  2 00:12:52.323: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessSocketExceptions: gConnTab=0x13C57EC0, a

ddr=80.90.160.221, port=5060, local_addr=10.0.128.11, connid=3, transport=UDP

Feb  2 00:12:52.323: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostCloseConnection: Posting UDP conn close for addr=80.90.160.221, port=5060, local_addr=10.0.128.11, connid=3

Feb  2 00:12:52.323: //-1/xxxxxxxxxxxx/SIP/Transport/sipDeleteConnInstance: Deleted conn=0x13C582A4, connid=3, addr=80.90.160.221, port=5060, local_addr=10.0.128.11, transport=UDP

Feb  2 00:12:52.323: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type rned: 2 for event 53

Feb  2 00:12:52.323: //-1/xxxxxxxxxxxx/SIP/Error/act_socket_send_msg_failure: Send Error to 80.90.160.221:5060 for transport UDP

Please do this test quickly! I need to sleep

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[moath.almogbel]

Hi aok,

I do appreciate your time and sorry for not replying earlier, but I have fallen sleep also as it was 3:00 AM here

I have updated the transport to tcp, and attached is the new debug.

Thanks alot Again!

[Ayodeji Okanlawon]

Moath good morning..Hope you had some sleep..

Ok here is what I think is going on..because I have not seen this before, I didnt spot it on time..

Your sip trunk is configured to use TCP as the transport protocol. However your inboud dial-peer on gateway is set to use UDP (the default)

Here is an inbound connection from CUCM to the gateway

Feb  2 09:38:32.766: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x1E80170, addr=10.1.3.12, port=53512, local_addr=10.0.128.11, connid=3, transport=TCP

Hence we need to match this on the inbound dial-peer..so configure dial-peer 10 as follows and test again..Please send the debugs too

dial-peer voice 10 voip

session transport tcp

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[moath.almogbel]

Good morning aok,

I have updated the dial-peer 10 to use tcp. attached is the debug.

One more thing, what if I changed the SIP trunk from CUCM to CUBE to use UDP instead of TCP, will this solve the problem? shall I try this?

[Ayodeji Okanlawon]

Moath, Lets hold on first and see whats happening. The logs you have sent is incomplete. Please send the full logs...

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[moath.almogbel]

Aok,

I have done two fresh tests, you can find them attached.

really appreciated.