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]

Yes, I had forgot that you told us the strange message about "time".

Now if you want to try to change silence suppression configure "no vad" under dial-peer and see if that helps.

[paolo bevilacqua]

Hi,

I was looking again at the trace. The call appears to be made to 11 digits number. I understand that Thailand uses 7 digits plus area code, else if dialing internationally you may need to prefix with 00.

[ngss]

Now we have a bit more explaination.

Caller Send INVITE

SER/SIP Proxy send 100

SIPPY (an IVR, I believe is *) send 183 Session Progress, along with early media (one-way audio) to announce to Caller that 'something is invalid', last about 6 seconds.

Then the caller hung up.

That's explain the time taken between 183 and CANCEL.

Now, let see if you sip-ua successfully registered with your provider,

Pls issue this command : sho sip-ua register status

Your debug trace shows that the caller still be allowed to send INVITE regardless the sip-ua register status.

To find out what actually happens to your original INVITE, debug ccsip message, ask the caller to hold the line even after the IVR message, I expect something like 4XX respone.

To solve issue, you need help from your provider to inform you what is not ok from your side.

Thanks

SS

[paolo bevilacqua]

Hi ngss,

as I was mentioning before, if the called number is actually the one present in initial trace, it doesn't make sense, as the ITSP appears to based in Thailand.

[Ph0kiszar]

Yes, some time users call to thailand number and international number, voice gw in thailand this sip server in singapore. till now they can't call to any destination and i'm tested "no vad" nothing difference. thank you so much.

[paolo bevilacqua]

Hi phokiszar, the thing is that being the sip server in singapore, you need to send all calls with 00 before the e.164 number, possibly only calls to singapore can be sent as national calls, but you should check this with the ITSP.

Please use an translation-profile to add 00 or tell you users to call with 00...

If you want to catch calls to Thailand and then add 00 and CC this is also possible, again using the translation-profiles.

Hope this helps, if so please rate post!

[Ph0kiszar]

I have tested the translation rule is the same. I'm talking with provider they give me some information, log from server just like this.

Calling-Station-Id = 'None'

May 11 11:45:48: Authorization failed: Failed - Invalid Account number

May 11 11:45:48: Authentication reject response

from the log above "Calling-Station-Id = 'None'" which command or parameter can make this field have calling-id?

thank you all.

[paolo bevilacqua]

Hello,

from the log, it seems that you are using "200' as username. However, the ITSP never challenges for authentication.

Is this "200" the username that the ITSP has given you? What have you configured as "authentication" under sip-ua ?

[Ph0kiszar]

Yes, the first time I use that number of dial-peer. now I have change for a while to sip number and tested then got log same above.

[paolo bevilacqua]

What I'm asking, is that the ISP should have give you username and password and possibly a realm, do you have configured that under sip-ua ?

[Ph0kiszar]

Now i'm done this case sip provider they optimize some thing in their sip server. Maybe authentication method.

Thank you so much p.bevilacqua and other.

[aboschetti]

Hi,

I've noted your specific competence into Unified Communications and sip configurations so I wish to post you a question.

I've to implement multiple sip registration with a sip provider using a voice gateway; I know that is accepted only 1 authentication for router.

How can I do ?

[paolo bevilacqua]

Unfortunately nothing . Submit your request to Cisco for future implementation. One possible contact is Tony Huynh <tonhuynh@cisco.com>, he is CME's TME.

[aboschetti]

Sip provider give me 5 accounts related to 5 Pstn numbers assigned to my profile.

Now I'm able to use only 1 number (the number specified into authentication username ..)

How can I use the others ?

I've also a problem with Dtmf on sip connections ..

[adrianic2003]

Hi Ph0kiszar,

I have an issue with my sip trunk. I'm using a CCME on 2800 router trying to register it with my ITSP using a SIP Trunk.

My configuration is:

!

dial-peer voice 800 voip

translation-profile outgoing strip-sip

destination-pattern 7[2-9]..[2-9]......

redirect ip2ip

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

acc-qos guaranteed-delay audio

!

!

!

sip-ua

authentication username xxx password xxx realm dns:sip.x.ca

no remote-party-id

retry invite 5

retry response 3

retry bye 5

retry cancel 5

retry prack 5

retry notify 4

retry register 5

retry options 5

timers connect 100

timers connection aging 30

timers register 600

registrar dns:nat.babytel.ca:5065 expires 3600

sip-server dns:sip.babytel.ca:5060

notify telephone-event max-duration 3000

!

and the outputs of the "debug ccsip messages" is:

Mar 23 17:40:38.386: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

REGISTER sip:nat.babytel.ca:5065 SIP/2.0

Via: SIP/2.0/UDP 192.168.5.240:5060;branch=z9hG4bK2BF6E1

From: <>14168486814@nat.babytel.ca>;tag=10A1A5C-1211

To: <>14168486814@nat.babytel.ca>

Date: Fri, 23 Mar 2007 17:40:38 gmt

Call-ID: 304915F7-D89B11DB-836EE10D-B064E7F7

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

Max-Forwards: 70

Timestamp: 1174671638

CSeq: 5 REGISTER

Contact: <14168486814>

Expires: 3600

Content-Length: 0

Mar 23 17:40:38.418: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 403 Forbidden (Outbound Proxy Policy)

To: <>14168486814@nat.babytel.ca>;tag=6bc3de4f

From: <>14168486814@nat.babytel.ca>;tag=10A1A5C-1211

Via: SIP/2.0/UDP 192.168.5.240:5060;branch=z9hG4bK2BF6E1

Call-ID: 304915F7-D89B11DB-836EE10D-B064E7F7

CSeq: 5 REGISTER

Server: DITC-PeerPoint C100/3-05-26-GA7p2

Content-Length: 0

based on your experience with sip trunk can you give me a hand to solv this problem please.

I would appreciate so much your help.

Thak you!

Adrian