problem with sip trunk

[Ph0kiszar]

I have sip account from provider and config to sip-ua with cisco 3800 series all peer behind my pbx are registered then I have call to some telephone number

I have hear from IVR of sip server "this's time number is not valid".

What's the "time number" that the sip server want? what command can solve this problem?

!

dial-peer voice 3 voip

destination-pattern T

redirect ip2ip

voice-class codec 1

voice-class sip transport switch udp tcp

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

acc-qos guaranteed-delay audio

!

!

sip-ua

authentication username **** password ****

no remote-party-id

retry invite 3

retry response 3

retry bye 3

retry cancel 3

retry register 10

timers connect 100

timers connection aging 30

mwi-server ***ip*** expires 3600 port 5060 transport udp unsolicited

registrar ***ip*** expires 3600

sip-server ***ip***

notify telephone-event max-duration 3000

!

Thank you.

[paolo bevilacqua]

Hi,

beside the fact that the destinatin pattern seems a little strange - just T, please look at "debug ccsip meesage" and "term mon" to see if the number you are sending is valid.

Hope this helps, please rate all useful posts!

[Ph0kiszar]

Yes, all digit are valid to sent. I have been debug ccsip all to check it before. I try troubleshooting this problem before I'm post here .

thank you p.bevilacqua.

[Ph0kiszar]

Has more any suggestion? I'm still try to solve this problem.

[paolo bevilacqua]

Can you send output of "debug ccsip message" ? If the called number is valid as you say, you should ask your provider why is not placing the call.

[Ph0kiszar]

I'm discuss with provider, they tell me and show the log in sip server. I see error with my account in billing system, my provider tell me some parameter or some thing about account not send to billing system but other it going fine. the problem in the billing how can I do with this problem? I just know only command about account of sip server "authentication username" under "sip-ua" .

[paolo bevilacqua]

Hello,

The SIP request that the cisco router makes is perfectly standard and it works with all service providers.

If your provider has problem with it, he should at least specify what exactly is wrong with the cisco and why. Unless we know this, there is nothing that can ba said about it.

[ngss]

Hi,

To solve this issue, you really need to show debug ccip message.

My suspicion is that your provider expect your gateway to be authenticated with them.

First, uou need verify that the authentication is ok during the REGISTERation process.

The ccsip message will display that.

Thanks

SS

[paolo bevilacqua]

Very likely is not registering, considering that "credentials" under sip-ua is not present in the configuration originally posted.

But many providers let place calls from unregistered users using http authentication.

And I was asking from "debug ccsip message" at my first post :)

[Ph0kiszar]

I'm sorry any one,I'm busy have no time to access this router. but I have now lets see debug message. thank you all.

[paolo bevilacqua]

Hi,

The ITSP fails to return status after the initial trying and session progress. It does not challenge for authentication.

I think I've seen this already in another case, but cannot remember what came out of it. Perhaps time to switch provider, there are many to choose from.

[ngss]

Hi,

The debug shows that after the gw receive the 183 Session Progress, it immediately send CANCEL

Apr 20 11:51:34.620: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 183 Session Progress

...........

Apr 20 11:51:42.940: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

CANCEL sip:1323442200.......

In the 183 response, there is also

a:silencesupp:off

This lead me to suspect the gateway CANCEL the INVITE because of this statement.

Now, basically you have your voice gateway, a SIP Proxy in between and PSTN Termination on the other side, such Softswitch or IP-PBX.

The SIP Proxy only convey back what it got from SS/IP-PBX.

Is this possible to ask your provider not to send a=silencesupp:off field?

If possible also, issue debug ccsip all to find out what is happening at the gateway. Take case not impact the user while doing this.

Thanks

SSng

[paolo bevilacqua]

Hi ngss,

I do not quite agree with your analysis of the debug. Cisco sends cancel 8 seconds after receiving session progress, not immediately after.

I believe this is due to calling user closing the call, due to nothing received. This can be confirmed by the original poster. In fact, the call should be left open until further messages are received from the ITSP, or a timeout occours.

Cisco should not have problems with no VAD and anyway when media negotiation fails, an error is thrown, not a cancel.

[ngss]

Hi,

Yeah, I wonder what happen between 183 and CANCEL process.

debug ccsip all may reveal us something.

In other case, removing the a=silenecsupp solve it.

The provider SIP proxy is SER. The 183 reply originally came from Softswitch or PSTN termination.

The trace at the SIP Proxy may help to find out what happens.

Thanks

SS

[Ph0kiszar]

from the debug ccsip message, users call to some destination and they're hear that IVR told "time number is not valid" then users will hang up immediatly because they're know can't call to destination.