CUBE for SIP Trunk (service provider) via SRST

[Gustavo Moreschi]

I have a 2921 router CUBE mode, and I 'm wondering to know how to setup the SRST if CUCM goes down. I do not have any FXO or T1/E1 to send directly to the PSTN. I know if the WAN goes down there is no backup, however if the CUCM goes down, I'd like to be able to make local, long distance calls in SRST mode SCCP through SIP Trunk with my service provider. Is it possible? If so, please explain how it will work. Thanks!

Service Provider - SIP Trunk - CUBE - SIP - CUCM

[Leo Salcie Tejeda]

Hi Gustavo,

If what you want to reach is to register the phone to the 2911 router hen CUCM goes down, you will need to use SRST.

You ca follow this guide by IPExpert:http://blog.ipexpert.com/2012/03/07/high-availability-series-1-srst-base-configuration/#more-10399

Regards

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[Gustavo Moreschi]

Hi, sorry I forgot telling that I have already setup the call-manager fallback, but even though the calls do not go to SIP Trunk (SP) when the Call Manager is down. I can see the IP Phone (SCCP) ephone registered in SRST, the correct voice-port, and dial-peer. However, it doesn't work. Here is the config, and what I found an error message based on the debug:

call-manager-fallback

secondary-dialtone 9

max-conferences 8 gain -6

transfer-system full-consult

timeouts interdigit 5

ip source-address 10.225.34.1 port 2000 strict-match

max-ephones 150

max-dn 300

system message primary SRST Aktiv - CUCM Unreachable

Received:

SIP/2.0 404 Not Found Reason: Q.850;cause=1

Any idea what it might be? Thanks!

[Ayodeji Okanlawon]

Can you send your full sh run. How are you routing calls to the PSTN during normal operation? Send us your full config and the full debug ccsip messages for a failed call

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[Gustavo Moreschi]

Here is the call flow:

IP Phone (SCCP) - SIP - CUCM - SIP - CUBE - SIP Trunk (Service Provider): works everything (normal operation)

IP Phone (SCCP) in SRST - CUBE - SIP Trunk (Service Provider): doesn't work any call in a SRST mode through SIP Trunk (SP)

Thank you!

[Ayodeji Okanlawon]

I have looked at your config and logs,,,

There is nothing wrong with the sip trunk in SRST except that it looks as if the number you are dialling is incorrect

"002232565809". When I dialled this number from my mobile phone I got number unobtainable. Your provider is saying the number cant be routed

Have you tried dialling this number in SRST "00393426272432"..This is a working number from your logs.

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[Gustavo Moreschi]

I created ACLs on extension 390 to test in SRST mode. This number is correct 002232565809 when I dial from Germany's office it works fine without SRST mode.

The calling number is: 32221090390, and the called numbers that I used to test are: 02232565809 or 069427299222 both did not work in SRST from the extension 390.

I have found based on the debugs SIP message, instead of getting "recevied SIP/2.0 183 session progress", it is getting "received: SIP/2.0 404 Not Found".

As I'm using early-offer forced without a MTP, it works when the CUCM is up, but once the CUCM is down, the phones get registered on SRST, but they cannot make any outbound calls through my SIP Trunk (service provider). I do not know if it relates to the MTP, or security somehow, or any other thing..., I just know for sure when those phones are in SRST, any outbound calls do not work.

Any thoughts is much appreciate. Thanks!

[Ayodeji Okanlawon]

There is no problem wit early offer. In SRST/CCME, the default is Early offer..And from the trace we can see this.. The invite was sent to your provider with SDP.

INVITE sip:002232565809@159.63.122.164:5060 SIP/2.0

Via: SIP/2.0/UDP 10.225.34.1:5060;branch=z9hG4bK17973B65

Remote-Party-ID: "390" <32221090390>;party=calling;screen=no;privacy=off

From: "390" <32221090390>;tag=1451FC8C-31A

To: <002232565809>

Date: Mon, 21 Jan 2013 14:20:28 gmt

Call-ID: 8A40C7CA-630C11E2-9E23A0FF-16692D8F@10.225.34.1

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 2243341544-1661735394-2652807423-0375991695

User-Agent: Cisco-SIPGateway/IOS-12.x

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

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1358778028

Contact: <32221090390>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 268

v=0

o=CiscoSystemsSIP-GW-UserAgent 7219 5369 IN IP4 10.225.34.1

s=SIP Call

c=IN IP4 10.225.34.1

t=0 0

m=audio 16758 RTP/AVP 0 101 19

c=IN IP4 10.225.34.1

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

a=ptime:20

I have looked at the traces for normal operation for calls coming from cucm and the SRST trace and I dont see any difference. You will need to contact your provider and find out why they are sending "404 Not Found" to you

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"opportunity is a haughty goddess who waste no time with those who are unprepared"