problem with sip trunk

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Apr 5th, 2007
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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.

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Overall Rating: 4 (2 ratings)
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paolo bevilacqua Thu, 04/05/2007 - 01:27
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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 Thu, 04/05/2007 - 02:16
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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 Fri, 04/06/2007 - 04:03
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Has more any suggestion? I'm still try to solve this problem.

paolo bevilacqua Fri, 04/06/2007 - 04:11
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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 Tue, 04/17/2007 - 23:54
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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 Wed, 04/18/2007 - 03:07
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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 Thu, 04/19/2007 - 00:15
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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 Thu, 04/19/2007 - 05:50
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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 Fri, 04/20/2007 - 04:15
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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.







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paolo bevilacqua Fri, 04/20/2007 - 04:25
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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 Fri, 04/20/2007 - 09:56
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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 Fri, 04/20/2007 - 10:24
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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 Fri, 04/20/2007 - 16:17
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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 Fri, 04/20/2007 - 20:27
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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.

paolo bevilacqua Sat, 04/21/2007 - 05:10
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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 Sat, 04/21/2007 - 06:24
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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 Mon, 04/23/2007 - 02:37
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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 Mon, 04/23/2007 - 02:57
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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 Mon, 04/23/2007 - 04:18
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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 Mon, 04/23/2007 - 04:58
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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 Thu, 05/10/2007 - 19:57
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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 Fri, 05/11/2007 - 03:18
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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 Mon, 05/14/2007 - 19:41
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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 Tue, 05/15/2007 - 03:44
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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 Tue, 05/15/2007 - 03:56
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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 Mon, 01/21/2008 - 02:17
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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 ?

aboschetti Mon, 01/21/2008 - 02:42
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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 Wed, 05/23/2007 - 13:47
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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: [email protected]>;tag=10A1A5C-1211

To: [email protected]>

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:

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: [email protected]>;tag=6bc3de4f

From: [email protected]>;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


paolo bevilacqua Wed, 05/23/2007 - 20:25
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Hi Adrian,


Looks like your NAT device is not fixing the contact address in the registration request, please check if you can have it do that.

Also please configure credentials under sip-ua for registration.

adrianic2003 Thu, 05/24/2007 - 06:08
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Hi p.bevilacqua,


Thank you for your fast reply.

I put the credentials on the "sip-ua", but I don't know how to fix the NAT problem. I have a PIX firewall in front of may device which is doing PAT from the inside network using the outside interface. When I talked to the ITSP they told me that my REGISTER request should go with: From: [email protected] and To: [email protected],


but in my case both "From" and "To" are going with [email protected]


I tried to change the values on "sip-ua" for "registrar" and "sip-server" to fix the problem but I didn't find the solution.

If you know what I have to change please let me know.

Thank you for your time again!


Adrian



paolo bevilacqua Thu, 05/24/2007 - 06:56
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Hi how have you set credentials ? I'm not sure the logic used for selecting the @part, how it works. And CCME is working with various provider without this problem.

Which IOS are you using ? can you try 12.(11)XJ3 that has lot of SIP improvements ?



adrianic2003 Thu, 05/24/2007 - 08:52
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Hi,


I have solved the problem with registration. I used the ip address insted of those names (nat.provider.ca..). Now my gateway is registered with their proxy server, and I can receive calls on that number. But I cannot make calls. When I'm trying to call a local number (for ex) using the dial-peer configured for sip I'm getting 403 Forbidden:


Mar 24 12:30:46.602: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:[email protected]:5065 SIP/2.0

Via: SIP/2.0/UDP 192.168.5.240:5060;branch=z9hG4bKB771D5F

From: ;tag=514C638-930

To: [email protected]>

Date: Sat, 24 Mar 2007 12:30:46 gmt

Call-ID: [email protected]

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 1335529150-3644461531-2287067405-2959402999

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

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

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1174739446

Contact:

Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=3000"

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 297


v=0

o=CiscoSystemsSIP-GW-UserAgent 5259 3281 IN IP4 192.168.5.240

s=SIP Call

c=IN IP4 192.168.5.240

t=0 0

m=audio 16608 RTP/AVP 18 101 19

c=IN IP4 192.168.5.240

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

a=ptime:20


Mar 24 12:30:46.634: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 403 Forbidden (Outbound Proxy Policy)

To: [email protected]>;tag=3e08ac65

From: ;tag=514C638-930

Via: SIP/2.0/UDP 192.168.5.240:5060;branch=z9hG4bKB771D5F

Call-ID: [email protected]

CSeq: 101 INVITE

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

Content-Length: 0


Now i'm working on this. If you have any suggestions please let me know.

Thank you so much!


Adrian

adrianic2003 Thu, 05/24/2007 - 09:49
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I found out what was the problem with outgoing calls. I have to manipulate the calling number that every outgoing call to go out with the calling number the ITSP assigned it to me.

Thanks for everyting!


Adrian

paolo bevilacqua Tue, 05/29/2007 - 07:53
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Glad to know this is also working.

Please remember to rate all useful posts using the scrollbox below!

adrianic2003 Thu, 06/21/2007 - 07:24
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Hi,


I'm facing dtmf-relay issues through sip trunk. There are some PBXs with IVR system wich does'n recognize my dtmf tones (when they ask to enter "1" for Sales - for examples). I have configured on my dial-peer like this:


dial-peer voice 810 voip

translation-profile outgoing strip-sip

service session

destination-pattern 71[2-9]..[2-9]......

redirect ip2ip

rtp payload-type nte 127

voice-class codec 1

session protocol sipv2

session target ipv4:216.18.125.7:5065

dtmf-relay rtp-nte sip-notify cisco-rtp


and my voice-cleass codec:


voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

codec preference 3 g729br8

codec preference 4 g711alaw


I would appreciate any suggestion for this issue.

Thank you!


Adrian


paolo bevilacqua Thu, 06/21/2007 - 07:34
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Hi Adrian,


I would proceed for steps. If the tones are never recognized by any system that you call via VOIP, then you may want to try changing the setting under DP 810, for example the order of dtmf-relay and the rtp payload type to default.


But, if some call can trasmit DTMF tones, and some does not, then you would need to find out first which numbers are not working, as the issue can be due to your call being terminated to different GW's depending on your TISP routing, in this case you might end with multiple DPs with different dtmf-relay settings, depending on the number called.


Hope this makes sense.

Alistair Pidd Thu, 06/26/2008 - 23:39
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Hi,


Did anyone get anything further with the DTMF tones through SIP? I have the same issues whereby all calls to mobiles and landlines are fine but calls to IVR's dont recognise the DTMF. We have a SIP trunk to a provider which then gets converted to H323 to the Call Manager. We have the following dial peers

dial-peer voice 100 voip

tone ringback alert-no-PI

description Inbound

session protocol sipv2

session target ipv4:10.0.220.45

incoming called-number .

dtmf-relay rtp-nte

no vad

!

dial-peer voice 101 voip

description Oubound

destination-pattern 9T

translate-outgoing called 1

session protocol sipv2

session target ipv4:10.0.220.45

dtmf-relay rtp-nte

no vad

aboschetti Fri, 06/27/2008 - 00:06
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Hi, you can modify the rtp payload.

I'd the same problem and I solved with these commands:


rtp payload-type cisco-codec-fax-ind 102

rtp payload-type nte 96

dtmf-relay rtp-nte

Alistair Pidd Fri, 06/27/2008 - 00:23
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Hi


thanks for the tip. However that doesnt seem to work either. I added it to my outbound dial peer to the service provider. The calls connect okay but DTMF doesnt work. If I push the calls out of our H323 to H323 gateway using alphanumeric and slow start then they work okay. This H323 to SIP gateway with fast start enabled wont pass the tones.

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