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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: April 24 2006 at 9:35am | IP Logged
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I have downloaded and tested your Media Engine product Dual Line IVR Server. The software work correctly and detects DTMF when I establish a call with SJPhone directly. But when I connect through asterisk SIP Proxy it does not detect DTMFs. I tested with both a SIP hardphone and SJPhone softphone.
I use following code to enable connecting to SIP proxy (asterisk SIP Proxy).
status = SipTelephonyEnable(pCPhoneBase->hSipEngine);
status = EnableSipDomain(pCPhoneBase->hSipEngine, "192.168.1.254"); if (status == SipSuccess) {
status = EnableSipProxyServer(pCPhoneBase->hSipEngine, "192.168.1.254",5060);
if (status == SipSuccess)
status = EnableSipRegisterServer(pCPhoneBase->hSipEngine,
"126",
TRUE,
"192.168.1.254",
5060,
8,
10,
5000,
TRUE);
}
I have three other questions :
1- Which of the following VOIP call managers LanScape VOIP Media Engine works perfectly (specially for conference and call transfer).
a) Mitel
b) Asterisk
c) Cisco
d) Alcatel
e) Siemens
2- Does it support Fax receiving and Fax Sending? If yes which image formats is supported ?
3- If we buy a 64 line support licenses, can we distribute as many softwares as we want without buying another license?
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: April 24 2006 at 12:10pm | IP Logged
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Hi Jalal,
About registering with Asterisk (*):
Your code looks OK. Is the IVR app successfully registering with Asterisk? If you want, post a VOIP Media Engine SIP log file so we can see exactly what is going on. Then we can assist you further.
The last parameter in your call to EnableSipRegisterServer() can be FALSE – you shouldn’t need auto NAT detect.
Assuming you are registering with * OK, make sure you perform test calls to the IVR server app using aLaw or uLaw (G7.11) codecs. The IVR server is performing in-band DTMF detection using the received RTP media stream.
Item 1 – Call Managers:
We only have information regarding *. We have not performed testing with the other vendor products you mentioned. However, the VOIP Media engine is based on rfc3261 so there should be a high degree of interoperability.
Call transfers:
Call transfer operations will be handled by * using BYE-Also method of call transfer (unattended call transfer). Asterisk supports this but will log an entry stating that REFER method should be used. We are working to implement the new REFER method in an upcoming release of the VOIP Media Engine.
Conference calling:
If you want * to perform the conference call function: Setup normal “static” conference room definitions in your * setup. All callers who want to take part in a conference session may call the proper * conference room number.
If you want the VOIP Media Engine to handle conferencing: Any VOIP application based on a multi-line version of the VOIP Media Engine has the ability to act as a conference point. For example, if you were to develop a soft phone, you would call all the conference parties and then punch the callers down into conference mode. All of the media for all callers will be relayed properly as to enable full multi-user conferencing.
Item 2 - Fax receiving and Fax Sending:
This is a good question and we have a number of options that we are considering for development. There are different ways to achieve this. At the moment, the VOIP Media Engine does not handle Fax directly but it can send/receive any G711 (aLaw/uLaw) fax modulated media stream over a VOIP network.
If you wanted to interface legacy stand alone fax machines to your VOIP media Engine developed application, you could do this by interfacing the FAX machine to your VOIP app using any G711 capable SIP ATA (analog telephone adaptor).
If you want to develop an application that can fax directly over VOIP, then you will need to develop or procure a fax software engine that has the ability to generate modulated digital fax data (that will be sent to the VOIP connection) and be able to decode modulated digital FAX data (that will be received from the VOIP connection).
Important: The amount of packet loss experienced when a FAX is transmitted/received over VOIP has a big influence on the success or failure of FAX over VOIP.
Item 3 –Licensing
If we buy a 64 line support licenses, can we distribute as many softwares as we want without buying another license?
Yes. Be aware that each license you purchase is for a single application you develop. For example: If you want to develop a soft phone and also develop a voice mail server, you must purchase a product license for each separate product (application) you develop. Once you develop your application(s), sell as many of them as you want. No per copy royalties – you pay LanScape just once.
Repost as required.
Support
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Pre-Sales Technical Support
Topic: Dual Line IVR Server DTMF Detection problem
