Configure DMTF conversion inband to Outband

[cmmfahim1]

Dear friends,

I have the following setup.

Voice mail server  <--> SIP proxy (VOIPswitch)  <---> Linksys ATA  <--> User

The SIP proxy support only DTMF inband and can't be changed since it is global config. But the voicemail server understands only the SIP DTMF outband.

What I'm trying to do is install a cisco voice router in between Voice mail server and SIP proxy (VOIPswitch) so that the final setup will be looks like

Voice mail server <-->Router  <--> SIP proxy (VOIPswitch)  <---> Linksys ATA  <-->User

When i have tried to implement the same with the cisco 7204 router i face the issue that there is no SIP notify option available in the DTMF-Relay type even though the ios version suggest it has.

Can you please help me on following areas and solve this difficult issue

1. Will this setup will work if i create two dial peers one for Voice mail server and the other one for SIP proxy (VOIPswitch)

2. How to enable the DTMF outband in the router under the dial peer since the sip notify is not available as supported type

3. I thought to create a dial peer for Voice mail server with destination pattern as 900 (access number) and create a dial peer for SIP Proxy server with destination pattern as .9T is this ok?

4. Any other suggestion or approach?

Thanks in advance and appreciate your quickest response.

Thanks again and Happy Cisco!

Fahim

[ADAM CRISP]

Hi Fahim.

DTMF SIP-Notify appears after you set the dial-peer's protocol to SIP.

But I think you're going to struggle to convert inband DTMF into out of band without any transcoding resource.

I am not familiar with VOIPswitch, however if it's just a proxy I doubt it will get involved in the SDP notifications of RFC2833 - RTPNTE availability.

I would recommend capturing the SIP dialogue either side of the VOIPswitch to confirm this.

With respect to the dial-peer for the VM server, why does this need a detination pattern, does it dial-out ?

Adam

[cmmfahim1]

Hi Adam,

First of all thank you very much for replying to my query. Please find my comments along with your points.

DTMF SIP-Notify appears after you set the dial-peer's protocol to SIP.

[FM] I have enabled session protocol sipv2. But still the dtmf-relay doend't show Sip notify or SIP-info. I think my router lacks with some DSP resources. what do you think?

But I think you're going to struggle to convert inband DTMF into out of band without any transcoding resource.

[FM] Do you mean DSP rsources?

I am not familiar with VOIPswitch, however if it's just a proxy I doubt it will get involved in the SDP notifications of RFC2833 - RTPNTE availability.

I would recommend capturing the SIP dialogue either side of the VOIPswitch to confirm this.

[FM] Do you mean DSP rsources?

With respect to the dial-peer for the VM server, why does this need a detination pattern, does it dial-out ?

[FM] Can you please help me on providing with sample dial peer config?

Thanks in advance

Fahim

[auppal]

The ios router does not need DSP's to convert 2833 to sip-notify. IOS router does need DSP's for inband dtmf i.e. within  the audio stream ( do not get confused by 2833 which is also in band) to sip-notify. All you need to make your case work is incoming dial peer with 2833 and the outgoing dial peer with sip-notify you should be good to go. If it doesnt work attach deb ccsip messages deb voip ccapi inout and deb voip rtp session named-event all in one file.

Thanks

AU

[ADAM CRISP]

Hi

As discussed attach your debug files if you can't get this working.

I've fired up a 7200 with 15.0(1)M2 and SIP notify is available, SIP INFO doesn't appear in this list.You don't need DSP for SIP Notofy to appear.

Yes transcoding resource = DSP = PVDM on ISR routers.

basic dial-peer would be..

dial-peer voice 1 voip
session protocol sipv2
  incoming called-number 900
dtmf-relay rtp-nte (I would be tempted to include other DTMF options here...)

dial-peer voice 2 voip
  session protocol sipv2
destination-pattern 900
  session target ipv4:
   dtmf-relay  sip-notify

This would cause the router to accept a call destined for "900" and then router the call to your VM machine.

Appologies if I causes confusion between inband audio and inband 2833

Adam

[cmmfahim1]

Hi Friends,

Sorry for late reply. I was on a field trip and couldn't access the web.

Thanks auppal & ADAM CRISP and I have learnt valuable points from your reply.

I have gone through all the comments and thanks for your valuable comments. The voice mail server seems to be connected with the SIP proxy using the proprietary method of access and den;t dial out any number. Hence the solution I have initially proposed seems to be irrelevant.

Since the setup is seems to be complex because of the nature of the voice mail server i'm trying to go for another type of soluton as below

1. ATA(RFC2833)------ --internet----->SIPproxy------->VoicemailServer
2. ATA(RFC2833)------ --internet----->SIPproxy------->3845router(Inband)---- --->3rdPartyCarri ers

earlier I have placed the router in between SIP proxy and VoicemailServer. Now decided to use the RC2833 at the ATA side so that voice mail server directly understand it and need to conver to inband before reaching the 3rd party carriers since most of them support only inband.

Hence the voice mail server issue sorted I need your help on configuring the router for all the carriers.

Can you please suggest any best solution to configure the router in shorter way rather than having dial peer profile for each carrier.

Thanks in advance

Fahim