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New Member

problem with sip trunk

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.

40 REPLIES
Hall of Fame Super Gold

Re: problem with sip trunk

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!

New Member

Re: problem with sip trunk

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.

New Member

Re: problem with sip trunk

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

Hall of Fame Super Gold

Re: problem with sip trunk

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.

New Member

Re: problem with sip trunk

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" .

Hall of Fame Super Gold

Re: problem with sip trunk

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.

New Member

Re: problem with sip trunk

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

Hall of Fame Super Gold

Re: problem with sip trunk

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 :)

New Member

Re: problem with sip trunk

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.

Hall of Fame Super Gold

Re: problem with sip trunk

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.

New Member

Re: problem with sip trunk

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

Hall of Fame Super Gold

Re: problem with sip trunk

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.

New Member

Re: problem with sip trunk

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

New Member

Re: problem with sip trunk

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.

Hall of Fame Super Gold

Re: problem with sip trunk

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.

Hall of Fame Super Gold

Re: problem with sip trunk

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.

New Member

Re: problem with sip trunk

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

Hall of Fame Super Gold

Re: problem with sip trunk

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.

New Member

Re: problem with sip trunk

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.

Hall of Fame Super Gold

Re: problem with sip trunk

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!

New Member

Re: problem with sip trunk

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.

Hall of Fame Super Gold

Re: problem with sip trunk

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 ?

New Member

Re: problem with sip trunk

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.

Hall of Fame Super Gold

Re: problem with sip trunk

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 ?

New Member

Re: problem with sip trunk

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.

New Member

Re: problem with sip trunk

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 ?

Hall of Fame Super Gold

Re: problem with sip trunk

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

New Member

Re: problem with sip trunk

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 ..

New Member

Re: problem with sip trunk

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

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