Cisco ATA (SPA112) and SIP trunks

[caknfd]

Have a multiple site system that was using 6 PRI's we recently removed the PRI's and went SIP. All is good except Cisco ATA's (SPA112) at remote sites now get reorder when dialing out.(No problem prior to the SIP conversion) They can dial 5 digit (internal numbers at any site) and when I routed a phone number out a local CO trunk that went OK.
We have some Polycom IP 6000 conference phones with the same COS, COR and SIP device capability that have no issues dialing out only the ATA's are having the issue.
Any thoughts?

All 3300's All Rel 6.0 SP3

 

[kwbMitel]

Were the ATA's in service prior to the change?

If not, the issue is likely with the dialing plan not being configured for ARS

**********************************************
What's most important is that you realise ... There is no spoon.

 

[caknfd]

The ATA's were there all along and worked fine on the PRI's only started to fail when I replaced the PRI's for SIP Trunks. Inbound is fine they just get reorder (fast busy when dialing out)

Craig

 

[Billz66]

perhaps routing issue?
if its sip trunks the ata will now be communicating with the exit point for the trunk

previously it would have been 'proxied' via the pabx if it had isdn trunks

If I never did anything I'd never done before , I'd never do anything.....

 

[kwbMitel]

To add to Billz66's suggestion

If this is routing through a MBG then the local networks need to be configured to allow the IP's on the endpoint devices.

**********************************************
What's most important is that you realise ... There is no spoon.

 

[caknfd]

The SIP trunks are on the local lan and not run through an MBG. Only the ATA's at the remote sites are having the issues all other digital, IP and SIP (Polycoms) are fine and all ATA's, Polycom's IP phones are are on the same subnet at each site so it is not a local network issue and that I would think would give me more of a one way audio issue and not a fast busy that I'm getting when dialing out on the ATA's would it not? I reconfigured one of the PRI's and in ARS routed mt office DID to go out the PRI and the ATA had no issue calling out that number so it is something between the SIP trunks and ATA configuration i'm just lost as to what it could be.

Craig

 

[Billz66]

can they recieve calls ?

it sounds to me like the signalling is trying to setup the call and then its giving re-order because it cant setup the speech path

signallying will be between ATA and pabx

speech will be ata to SIp provider

If I never did anything I'd never done before , I'd never do anything.....

 

[sarond]

Can you perform a SIP trace on the 3300 to see what message is being sent back to the SPA112.

Sometimes sending incorrect caller id will also stop an outbound call. Maybe try CPN Substitution.

As Billz66 has already asked, can you receive calls?

 

[caknfd]

Can receive calls.
Can make internal call to any site.
Can make external calls out a copper trunk on the local (to the ATA) 3300.
Can make external calls out a PRI on the same PBX as the SIP trunks.(remote 3300)
The caller ID that is put out is the Default CPN I see it when calling out via the PRI
These all were working fine until I replaced the PRI's with the SIP Trunking (The site also has a Polycom IP6000 same cor,cos and SIP device capabilities and it is working only the ATA's are not)
I did a SIP TCPDMP on the local 3300 and see I'm receiving a 488 not acceptable here message. In the attached trace the call to the 9434 number completed out the PRI at the main site the call ending in 4000 failed routing out the SIP trunk at the main site.
SIP is not something I am very strong in so if any of you would mind taking a look I would appreciate it.

Site doesn't have software assurance yet so I can't open a ticket with support but should soon.

Craig

 

[kwbMitel]

You could talk to the SIP provider and see if they are rejecting the call and why.

**********************************************
What's most important is that you realise ... There is no spoon.

 

[wireman50]

Its a client error as reported in the trace 488 points to a codec error from the ATA to the SIP provider. Can't tell you how to do that oon the Cisco though sorry. Just make sure that somewhere all the codecs that the cisco supports are enabled. Or use g729 as a start.

 

[caknfd]

Problem Solved!!!
Ran the SIP TCPDMP on the SIP Trunk side and found the call same 488 message but saw different responses in wireshark then on the ATA trace and it pointed me to Packetization Rate.
SIP trunks were set to 20ms found in the ATA they default to 30 or 0.030 so I changed the ATA to 0.020 and the calls started going through.

Thank You all for taking the time to look at this and assisting me.

Session Initiation Protocol (488)
Status-Line: SIP/2.0 488 Not Acceptable Here
Status-Code: 488
[Resent Packet: False]
[Request Frame: 10]
[Response Time (ms): 40]
Message Header
Via: SIP/2.0/UDP 10.52.2.185:5060;branch=z9hG4bK3717607792-103420180
To: <sip:17147544000@10.52.2.200>;tag=d06f4d7e
From: "Purchasing FAX"<sip:9515717538@10.52.2.185>;tag=0_3717607792-103420181
Call-ID: 3717607792-103420179
CSeq: 1 INVITE
User-Agent: ESBC10K-MDX-2.0.13.0-Build5
Warning: 304 8448-4005-5091-8150 "No supported packetization time"
Content-Length: 0

 

[kwbMitel]

Damn, I ran into that once, that's why we have a standard upload for our devices for config.

**********************************************
What's most important is that you realise ... There is no spoon.