ITSP to Speech Server 2007

EA · Post by EA » Wed Oct 15, 2008 2:52 pm

1. Brekeke Product Name and version: SIP Server 2.2

2. Java version:

3. OS type and the version: Vista

4. UA (phone), gateway or other hardware/software involved: voiptalk.org ITSP

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : pattern 8

6. Your problem:

I am new to VOIP and MS Speech Server 2007. My goal is to route an inbound call on a pstn virtual number (registered at voiptalk.org) through to MS Speech Server which will run a simple speech app. I have been struggling for the last few days to get this working with no success. I am looking to use Brekeke SIP Server to perform the UDP to TCP conversion as I am hoping this is the only barrier to the scenario above.

I was hoping it is as simple as:
ITSP -> Brekeke SIP Server -> Speech Server 2007

Question1 : Is it possible to simply do this or is there somthing I am missing?

I have the ITSP set to "Route to a PBX" which i have set as the IP of the machine hosting SIP Server.

Speech server is on the same machine but using port 6060.

- i can successfully make a local xlite call with transport=UDP which i can see SIP Server collecting, translating to TCP and mapping to port 6060 and subsequently hear the speech application.

- I can also make a phone call to my number and, using wireshark, can see the SIP connection request arriving at the SIP Server machine (192.168.1.64) . i just hear dead air on the line so I believe SIP Server is not picking up this request. No calls are logged in SIP server.

Source : 77.240.48.214
Destination : 192.168.1.64
Protocol : SIP/SDP
Info : Request: INVITE sip:01737888999@192.168.1.64, with session description

I have the following set in my dial plan:
Matching Patterns:
$request=^INVITE
$registered=false
$outbound=false

Deploy Patterns:
$transport=TCP
$auth=false
&net.sip.transport.follow.request=true
To=1234@127.0.0.1:6060

Question2: Can anyone tell me what is wrong with my setup above?

Many thanks is advance.

Harold · Post by **Harold** » Thu Oct 16, 2008 1:40 am

Can you make a call from the ITSP to your X-Lite?

> $outbound=false
remove this line.

EA · Post by EA » Thu Oct 16, 2008 11:54 am

yes, if i adjust my ITSP to 'route to ID' and setup xlite to register with the ITSP appropriately then I can successfully receive a call.

I also tried removing the $outbound as you suggested but with no success.

anyone else have any ideas? do i need to get SIP Server to register to the ITSP somehow?

Note: i am a little concerned that i am 'routing to PBX' at my ITSP but dont actually have a PBX (just SIP SERVER). is my original thinking flawed?

lakeview · Post by **lakeview** » Thu Oct 16, 2008 6:03 pm

> I can also make a phone call to my number and, using wireshark, can see the SIP connection request arriving at the SIP Server machine (192.168.1.64) . i just hear dead air on the line so I believe SIP Server is not picking up this request. No calls are logged in SIP server.

Which SIP response packet was returned to the ITSP at that time?

>Source : 77.240.48.214
> Destination : 192.168.1.64

It seems your SIP Server is behind NAT.
Did you set the NAT's global IP address at the SIP Server's settings?
[Configuration] -> [System] -> [Network] -> [Interface address 1]

EA · Post by EA » Mon Oct 20, 2008 1:41 pm

hi Lakeview,
sorry for the delayed response.

Regaring NAT, i did set the external ip of the router in the Interface Address but it makes no difference.

Regarding response packet, I'm not entirely sure how to view the response packet, could you give me brief instructions?
I clicked follow UDP stream, im not sure if what you are looking for, below is an extract:

INVITE sip:01737888309@192.168.1.64 SIP/2.0

Via: SIP/2.0/UDP 217.14.138.16:5060;branch=z9hG4bK511dd7f7;rport

From: "01737764019" <sip:01737764019@217.14.138.16>;tag=as2c19305e

To: <sip:01737888309@192.168.1.64>

Contact: <sip:01737764019@217.14.138.16>

Call-ID: 4f0b06ff75a995b513367fd8264eca5b@217.14.138.16

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Mon, 20 Oct 2008 20:21:56 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Type: application/sdp

Content-Length: 549

v=0

o=root 13381 13381 IN IP4 217.14.138.16

s=session

c=IN IP4 217.14.138.16

t=0 0

m=audio 15770 RTP/AVP 8 0 3 97 7 110 5 10 18 112 111 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 iLBC/8000

a=fmtp:97 mode=30

a=rtpmap:7 LPC/8000

a=rtpmap:110 speex/8000

a=rtpmap:5 DVI4/8000

a=rtpmap:10 L16/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:112 AAL2-G726-32/8000

a=rtpmap:111 G726-32/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

INVITE sip:01737888309@192.168.1.64 SIP/2.0

Via: SIP/2.0/UDP 217.14.138.16:5060;branch=z9hG4bK511dd7f7;rport

From: "01737764019" <sip:01737764019@217.14.138.16>;tag=as2c19305e

To: <sip:01737888309@192.168.1.64>

Contact: <sip:01737764019@217.14.138.16>

Call-ID: 4f0b06ff75a995b513367fd8264eca5b@217.14.138.16

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Mon, 20 Oct 2008 20:21:56 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces....

the one thing i noticed is that it says User-Agent: Asterisk PBX. I dont have an asteriks PBX so im not sure what that means.

lakeview · Post by **lakeview** » Tue Oct 21, 2008 11:05 am

Hi

>> the one thing i noticed is that it says User-Agent: Asterisk PBX.

It means your ITSP is using Asterisk..

The packet you pasted is an INVITE request packet..

Can you see any response packet after the INVITE request?
If so, please paste the response packet here.
The response packet will provide a clue.

EA · Post by EA » Wed Oct 22, 2008 12:58 am

Hi Lakeview

All i see is a series of invite rows,

INVITE etc etc etc
INVITE etc etc etc
INVITE etc etc etc
.
.
.

eventually after about 30 seconds the repeating INVITES stop and are followed by CANCEL's (about 8 of them)

lakeview · Post by **lakeview** » Wed Oct 22, 2008 10:41 am

Check the PC's firewall settings.

If a firewall is working on the PC, you need to open UDP port 5060 or disable the firewall.

EA · Post by EA » Wed Oct 22, 2008 2:45 pm

unfortunately that made no difference.

Is there any way to up the SIP Server logging so that i can see whether or not SIP server is "seeing" the invite?

lakeview · Post by **lakeview** » Wed Oct 22, 2008 3:02 pm

Can you see any session on the [Active Sessions] page?