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.
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.