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 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]

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: ***@nat.provider.ca and To: ***@sip.provider.ca,

but in my case both "From" and "To" are going with ***@nat.provider.ca

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]

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]

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:4163853924@nat.babytel.ca:5065 SIP/2.0

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

From: <204>;tag=514C638-930

To: <>4163853924@nat.babytel.ca>

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

Call-ID: 5342D608-D93A11DB-8857E10D-B064E7F7@192.168.5.240

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: <204>

Call-Info: <192.168.5.240:5060>;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: <>4163853924@nat.babytel.ca>;tag=3e08ac65

From: <204>;tag=514C638-930

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

Call-ID: 5342D608-D93A11DB-8857E10D-B064E7F7@192.168.5.240

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]

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]

Glad to know this is also working.

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

[adrianic2003]

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]

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]

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]

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]

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.