Hi,
 
We have a scenario where the call is coming directly over VoIP on the Cisco Platform. 
 
In normal cases, the call comes in as PSTN and then we forward the request to the VXML server to fetch the app. But here the call is a direct SIP Call to the Cisco router.
 
Is there a way to configure a dial-peer to process the SIP call directly? (Sorry, if this a very basic question)
 
Thanks
Anupam

Hi Anusha,

Thanks for your quick reply. So a Cisco router (such as 3945) can indeed receive a SIP call provided I configure the dial-peer correctly? So, if the IP address of the router is 192.168.10.10, I can make a call to the router using SIP:4680@192.168.10.10:5060 directly?

where 4680 is the DID of the voice application that it will map to (based on the dial-peer config)

In this scenario, I will not be needing any E1 trunk right?

Thanks,
Anupam

Thanks Anusha. That helps a great deal.

Regards
Anupam

HI Anupam,

When configuring the VOIP dial-peers there is no need for the DID and port configurations rather it should be as follows

dial-peer voice 2 voip
service myapp
destination-pattern .T
session-target ipv4:10.0.0.4
dtmf-relay xxxx


Where the destination-pattern and session target refer as following

destination-pattern 919…….
(calls with the digits 919+7 more digits will match this dial-peer)
session target ipv4:192.168.1.1
( the VOIP call will be sent to 192.168.1.1 when dial-peer is matched)

Thanks,
Anusha

Hi Anupam,

Yes you are right. For more clarity here is an example. Assume the call is coming from 3100 to 7100 and you are trying to load the application in the router where 7 series phones are connected, then the configuration will be as follows
!
dial-peer voice 3 voip
description "Pointing to SIP Dial-peer in 2800"
destination-pattern 31..
session protocol sipv2
session target ipv4:10.78.236.53
incoming called-number [6,7]1..
dtmf-relay sip-notify
codec g711ulaw
!

For more details you could refer to the dial-peer configuration in the following location
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/dial_peer/dp_confg.html#wp1066941

Thanks,
Anusha

Hi Anusha

With the above setup, when we make a SIP call to the Cisco router from a soft phone, we get back 488 (Not Acceptable Media). We were expecting it to be a codec issue and tweaked the config as follows but still getting a 488.

dial-peer inbound selection sip-trunk
sip-ua
g729-annexb override


Also, here is the trace of INVITE and the response:


INVITE sip:4000@203.122.28.219:5060 SIP/2.0
Via: SIP/2.0/UDP 9.126.106.243;branch=z9hG4bK097e6af30000003e5040841500000b9f0000000c;rport
From: "unknown" <sip:9.126.106.243>;tag=3d6c2e36bfef
To: <sip:4000@203.122.28.219:5060>
Contact: <sip:9.126.106.243>
Call-ID: 14C013012D79467883EA43B1DD6F64300x097e6af3
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 368
Content-Type: application/sdp
Supported: replaces,norefersub,timer
v=0
o=- 3555394197 3555394197 IN IP4 9.126.106.243
s=SJphone
c=IN IP4 9.126.106.243
t=0 0
m=audio 49164 RTP/AVP 3 97 98 8 0 101
c=IN IP4 9.126.106.243
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:active
a=sendrecv


SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 9.126.106.243:5060;branch=z9hG4bK097e6af30000003e5040841500000b9f0000000c;rport
From: "unknown" <sip:9.126.106.243>;tag=3d6c2e36bfef
To: <sip:4000@203.122.28.219:5060>;tag=EC2C840-563
Date: Fri, 31 Aug 2012 09:52:04 GMT
Call-ID: 14C013012D79467883EA43B1DD6F64300x097e6af3
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=65
Content-Length: 0

Can you please help us out here? (Sorry if this is a broad query - but if you can guide us in a direction that we can explore, that'll be great) Thanks!

Regards
Anupam

Hi Anusha,

Thanks for the suggestion. After trying the above command and strudying the logs etc. we figured out that the SIP phone that we are using to test does not support G729 but supports G711. So we made the folowing addition in the config now:


voice class codec 1
codec preference 1 g711ulaw


But we still get a 488 back when we make a call. Here's the INVITE and the response.

Received:
INVITE sip:4000@203.122.28.219:5060 SIP/2.0
Via: SIP/2.0/UDP 122.248.183.1:44301;branch=z9hG4bK097e6af3000000185044404100002ffe00000000;rport
From: "unknown" <sip:9.126.106.243>;tag=2cf42bc1158
To: <sip:4000@203.122.28.219:5060>
Contact: <sip:122.248.183.1:44301>
Call-ID: BE0AC393EF6E467BB0604E8244F12F3E0x097e6af3
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 368
Content-Type: application/sdp
Supported: replaces,norefersub,timer

v=0
o=- 3555638977 3555638977 IN IP4 122.248.183.1
s=SJphone
c=IN IP4 122.248.183.1
t=0 0
m=audio 30142 RTP/AVP 3 97 98 8 0 101
c=IN IP4 122.248.183.1
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:active
a=sendrecv

036262: *Sep 3 05:51:42.802: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 122.248.183.1:44301;branch=z9hG4bK097e6af3000000185044404100002ffe00000000;rport
From: "unknown" <sip:9.126.106.243>;tag=2cf42bc1158
To: <sip:4000@203.122.28.219:5060>;tag=1D59CEA8-2695
Date: Mon, 03 Sep 2012 05:51:42 GMT
Call-ID: BE0AC393EF6E467BB0604E8244F12F3E0x097e6af3
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=65
Content-Length: 0


Can you please give us a clue on what could be wrong here (I guess there might be something with the config that we still need to tweak)

Thanks
Anupam

Hi Anusha,

Thanks again for your prompt response.

This is the IOS version output from 'show ver':
Cisco IOS Software, 2800 Software (C2800NM-ADVIPSERVICESK9-M), Version 12.4(15)T8, RELEASE SOFTWARE (fc3)

Also, in the meanwhile, I added the following line to the dial-peer:
voice-class codec 1

and the call seems to be going through now (as in, it does not throw a 488)

But still, we are not able to hear the audio. All we are doing in the VXML is playing a wav file and exiting. (This vxml is constructed fine since we have already tested it with another instance of a Cisco router sometime back)

From the logs, it seems that the file did play and the application exited properly but we cant hear anything. I suspect it may still be a config issue but couldnt figure it out as yet.

Thanks
Anupam

Hi Anushka,

Thanks for helping out on this. Sent you the logs and configuration on email.

Regards
Anupam

Hi Anusha,

I added the following

session protocol sipv2
dtmf-relay sip-notify

in the dial-peer but the problem still persists :-( Cant hear anything on the call.

The dial-peer now looks like the following:

dial-peer voice 2 voip
service myapp
destination-pattern 4000
voice-class codec 1
session protocol sipv2
session target ipv4:203.122.28.219
incoming called-number 4000
dtmf-relay sip-notify

Thanks
Anupam