SIP 302 Move temporary failed

[desteh1]

Hi All,

i`m i doubt that the following Sip message we are receiving from an Endpoint, connected via a Sip Trunk is right.

For me it seems there are Information missing.

Let me explain the traffic case

An Attandant console is connected via a Sip Trunk at the CUCM system that is  running version 8.5.

In the case these Attandant console is switched to night mode and an internal cisco extension calling the internal or external queue number the call goes fail. ( Same happen when i calling from an external phone )

So we have these call flow

cisco Phone ---->  CtiRP Number 8890 ---> Sip Trunk ----> Attendant Server ----> Night extension is 1221

The Sip signalling flow is right, we receiviing a SIP Message 302 Move Temporary with a contact address

1221@10.16.1.30 but not part of the content is the reason.

For me, the information regarding Diversion reason are missing but mandatory adn so the call goes fail

I would like to know your opinion if i`m right with my assumption or not.

Here the mentined SIP message.

SIP/2.0 302 Moved Temporarily

Via: SIP/2.0/UDP 10.16.2.30:5060;branch=z9hG4bKa7eaad3d04de73

From: <sip:00369569695260@10.16.2.30>;tag=71330652~42c66868-dbc5-40eb-bd1a-d30bb01b6fe7-43409571

Call-ID: 93f9a880-2b11839e-5dba9b-1e02100a@10.16.2.30

CSeq: 101 INVITE

Server: Aastra NeTS 5.6.6.0/1.5.4.19

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

Supported: histinfo, replaces

Content-Length: 0

Contact: <sip:1221@10.16.1.30;maddr=10.16.1.30>

To: <sip:8990@172.16.1.61>;tag=390634443

Best regards

Stefan

[Hermanus Janse van Vuuren]

Are you dialing via a cisco Voice GW?

Have you got the following if you do have a gw?

voice service voip

no notify redirect ip2ip

allow-connections sip to sip

allow-connections sip to h323

allow-connections h323 to h323

allow-connections h323 to sip

[desteh1]

Hi,

we do not use a gateway at this traffic case. It is a pure CUCM sip Trunk Attendant server connection.

The call fails with the SIP message 503 Service unavailable.

Here the content.

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 10.16.2.30:5060;branch=z9hG4bKa7eaae23b23e62

From: <00369569695260>;tag=71330652~42c66868-dbc5-40eb-bd1a-d30bb01b6fe7-43409571

To: <8990>;tag=588532863

Date: Wed, 18 Dec 2013 11:14:38 GMT

Call-ID: 93f9a880-2b11839e-5dba9b-1e02100a@10.16.2.30

CSeq: 102 INVITE

Allow-Events: presence

Warning: 399 tk-cucm-rz "Unable to find a device handler for the request received on port 5060 from 10.16.2.30"

Content-Length: 0

Regards

Stefan

[Hermanus Janse van Vuuren]

error 503

This occurs when the source IP address that is sending the SIP INVITE  does not exist as a SIP trunk in CUCM.  You may have never created it,  or the IP address may not be the one that is in use

Check

Run On All Active Unified CM Nodes

On the SIP Trunk

[Manish Prasad]

Check this thread this might help you. --https://supportforums.cisco.com/thread/276600

Rate all the helpful post.

Thanks

Manish

[desteh1]

Hi,

i have checked the mentioned parameter and these parameter is already checked.

Deleting and reconfiguring the Sip Trunk does not solve the issue.

I`m getting all the time, at the specific traffic case the message 503 server unavailable.

At the cucm trace i can find the following,

|1,100,220,1.548666^10.16.2.30^*

15:24:50.202 |//SIP/SIPHandler/ccbId=0/scbId=0/findTrunkInfoByAddr: Cannot find the SIP Trunk with Name=10.16.2.30, Source Port=5060, IpAddress Type=0|1,100,220,1.548666^10.16.2.30^*

15:24:50.202 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer:  type=SIP_TIMER_TRYING value=500  retries=6|1,100,220,1.548666^10.16.2.30^*

15:24:50.202  |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_EXPIRES  value=180000 retries=0|1,100,220,1.548666^10.16.2.30^*

15:24:50.202  |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer:  type=SIP_TIMER_EXPIRES value=180000  retries=0|1,100,220,1.548666^10.16.2.30^*

15:24:50.202 |EnvProcessUdpHandler::handle_input - handle = 322|*^*^*

15:24:50.202 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 0|*^*^*

15:24:50.202 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 504 from 10.16.2.30:[5060]:

Why the system tries to find an Sip Trunk instead of routing the call at the SIP Phone 8961 with number 1903?

Any hints?

Best regards

Stefan